Merge tag 'asoc-v5.0' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v5.0/v4.20

As ever there's a lot of small and driver specific changes going on
here, but we do also have some relatively large changes in the core
thanks to the hard work of Charles and Morimoto-san:

 - More component transitions from Morimoto-san, I think we're about
   finished with this.  Thanks for all the hard work!
 - Morimoto-san also added a bunch of for_each_foo macros
 - A bunch of cleanups and fixes for DAPM from Charles.
 - MCLK support for several different devices, including CS42L51, STM32
   SAI, and MAX98373.
 - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
   MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
   TI PCM3060.

147 files changed, 8013 insertions, 1487 deletions

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 5fbd47a9177e..28867732a318 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -31,7 +31,6 @@ endif	# SND_ARM
 
 config SND_PXA2XX_LIB
 	tristate
-	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
 	select SND_DMAENGINE_PCM
 
 config SND_PXA2XX_LIB_AC97
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 2222b47d4ec4..9174f1b3a987 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -81,6 +81,12 @@ static int hda_codec_driver_probe(struct device *dev)
 	hda_codec_patch_t patch;
 	int err;
 
+	if (codec->bus->core.ext_ops) {
+		if (WARN_ON(!codec->bus->core.ext_ops->hdev_attach))
+			return -EINVAL;
+		return codec->bus->core.ext_ops->hdev_attach(&codec->core);
+	}
+
 	if (WARN_ON(!codec->preset))
 		return -EINVAL;
 
@@ -134,6 +140,12 @@ static int hda_codec_driver_remove(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
 
+	if (codec->bus->core.ext_ops) {
+		if (WARN_ON(!codec->bus->core.ext_ops->hdev_detach))
+			return -EINVAL;
+		return codec->bus->core.ext_ops->hdev_detach(&codec->core);
+	}
+
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
 	snd_hda_codec_cleanup_for_unbind(codec);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index f0ef52eb22a9..0a24037184c3 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -31,6 +31,7 @@
 #include <linux/types.h>
 #include <linux/io.h>
 #include <linux/pci.h>
+#include <asm/io.h>
 #include <sound/core.h>
 #include <sound/hda_codec.h>
 #include "hda_local.h"
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index 8e3275a96a82..3f813ea5210a 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -42,7 +42,7 @@
 #include "../codecs/da7219.h"
 #include "../codecs/da7219-aad.h"
 
-#define CZ_PLAT_CLK 25000000
+#define CZ_PLAT_CLK 48000000
 #define DUAL_CHANNEL		2
 
 static struct snd_soc_jack cz_jack;
@@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 	da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks");
 
 	ret = snd_soc_card_jack_new(card, "Headset Jack",
-				SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
+				SND_JACK_HEADSET | SND_JACK_LINEOUT |
 				SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 				SND_JACK_BTN_2 | SND_JACK_BTN_3,
 				&cz_jack, NULL, 0);
@@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
 	.mask = 0,
 };
 
-static int cz_da7219_startup(struct snd_pcm_substream *substream)
+static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &constraints_rates);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	machine->play_i2s_instance = I2S_SP_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL1;
 	return da7219_clk_enable(substream);
 }
@@ -162,11 +183,22 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_max_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->play_i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
@@ -177,21 +209,43 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream)
 
 static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_BT_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_BT_INSTANCE;
 	return da7219_clk_enable(substream);
 }
 
 static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
-	machine->i2s_instance = I2S_SP_INSTANCE;
+	/*
+	 * On this platform for PCM device we support stereo
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL0;
 	return da7219_clk_enable(substream);
 }
@@ -201,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream)
 	da7219_clk_disable();
 }
 
+static const struct snd_soc_ops cz_da7219_play_ops = {
+	.startup = cz_da7219_play_startup,
+	.shutdown = cz_da7219_shutdown,
+};
+
 static const struct snd_soc_ops cz_da7219_cap_ops = {
-	.startup = cz_da7219_startup,
+	.startup = cz_da7219_cap_startup,
 	.shutdown = cz_da7219_shutdown,
 };
 
@@ -233,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
-		.ops = &cz_da7219_cap_ops,
+		.ops = &cz_da7219_play_ops,
 	},
 	{
 		.name = "amd-da7219-cap",
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 77b265bd0505..cdebab2f8ce5 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -867,8 +867,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 
 	if (pinfo) {
-		rtd->i2s_instance = pinfo->i2s_instance;
-		rtd->capture_channel = pinfo->capture_channel;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			rtd->i2s_instance = pinfo->play_i2s_instance;
+		} else {
+			rtd->i2s_instance = pinfo->cap_i2s_instance;
+			rtd->capture_channel = pinfo->capture_channel;
+		}
 	}
 	if (adata->asic_type == CHIP_STONEY) {
 		val = acp_reg_read(adata->acp_mmio,
@@ -1036,16 +1040,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream)
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		period_bytes = frames_to_bytes(runtime, runtime->period_size);
-		dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
-		if (dscr == rtd->dma_dscr_idx_1)
-			pos = period_bytes;
-		else
-			pos = 0;
 		bytescount = acp_get_byte_count(rtd);
-		if (bytescount > rtd->bytescount)
+		if (bytescount >= rtd->bytescount)
 			bytescount -= rtd->bytescount;
-		delay = do_div(bytescount, period_bytes);
-		runtime->delay = bytes_to_frames(runtime, delay);
+		if (bytescount < period_bytes) {
+			pos = 0;
+		} else {
+			dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr);
+			if (dscr == rtd->dma_dscr_idx_1)
+				pos = period_bytes;
+			else
+				pos = 0;
+		}
+		if (bytescount > 0) {
+			delay = do_div(bytescount, period_bytes);
+			runtime->delay = bytes_to_frames(runtime, delay);
+		}
 	} else {
 		buffersize = frames_to_bytes(runtime, runtime->buffer_size);
 		bytescount = acp_get_byte_count(rtd);
diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h
index be3963e8f4fa..dbbb1a85638d 100644
--- a/sound/soc/amd/acp.h
+++ b/sound/soc/amd/acp.h
@@ -158,7 +158,8 @@ struct audio_drv_data {
  * and dma driver
  */
 struct acp_platform_info {
-	u16 i2s_instance;
+	u16 play_i2s_instance;
+	u16 cap_i2s_instance;
 	u16 capture_channel;
 };
 
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 64b784e96f84..64f86f0b87e5 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -97,4 +97,16 @@ config SND_ATMEL_SOC_I2S
 	help
 	  Say Y or M if you want to add support for Atmel ASoc driver for boards
 	  using I2S.
+
+config SND_SOC_MIKROE_PROTO
+	tristate "Support for Mikroe-PROTO board"
+	depends on OF
+	depends on SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8731
+	help
+	  Say Y or M if you want to add support for MikroElektronika PROTO Audio
+	  Board. This board contains the WM8731 codec, which can be configured
+	  using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins.
+	  Both playback and capture are supported.
+
 endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index cd87cb4bcff5..9f41bfa0fea3 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -17,6 +17,7 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
 snd-atmel-soc-classd-objs := atmel-classd.o
 snd-atmel-soc-pdmic-objs := atmel-pdmic.o
 snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o
+snd-soc-mikroe-proto-objs := mikroe-proto.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
@@ -24,3 +25,4 @@ obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
 obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
 obj-$(CONFIG_SND_ATMEL_SOC_PDMIC) += snd-atmel-soc-pdmic.o
 obj-$(CONFIG_SND_ATMEL_SOC_TSE850_PCM5142) += snd-atmel-soc-tse850-pcm5142.o
+obj-$(CONFIG_SND_SOC_MIKROE_PROTO) += snd-soc-mikroe-proto.o
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d3b69682d9c2..6291ec7f9dd6 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev)
 	struct ssc_device *ssc = dev_get_drvdata(dev);
 	int ret;
 
-	ret = snd_soc_register_component(dev, &atmel_ssc_component,
+	ret = devm_snd_soc_register_component(dev, &atmel_ssc_component,
 					 &atmel_ssc_dai, 1);
 	if (ret) {
 		dev_err(dev, "Could not register DAI: %d\n", ret);
-		goto err;
+		return ret;
 	}
 
 	if (ssc->pdata->use_dma)
@@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev)
 
 	if (ret) {
 		dev_err(dev, "Could not register PCM: %d\n", ret);
-		goto err_unregister_dai;
+		return ret;
 	}
 
 	return 0;
-
-err_unregister_dai:
-	snd_soc_unregister_component(dev);
-err:
-	return ret;
 }
 
 static void asoc_ssc_exit(struct device *dev)
@@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev)
 		atmel_pcm_dma_platform_unregister(dev);
 	else
 		atmel_pcm_pdc_platform_unregister(dev);
-
-	snd_soc_unregister_component(dev);
 }
 
 /**
diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c
new file mode 100644
index 000000000000..d47aaa5bf75a
--- /dev/null
+++ b/sound/soc/atmel/mikroe-proto.c
@@ -0,0 +1,165 @@
+/*
+ * ASoC driver for PROTO AudioCODEC (with a WM8731)
+ *
+ * Author:      Florian Meier, <koalo@koalo.de>
+ *	      Copyright 2013
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8731.h"
+
+#define XTAL_RATE 12288000	/* This is fixed on this board */
+
+static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	/* Set proto sysclk */
+	int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+					 XTAL_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(card->dev, "Failed to set WM8731 SYSCLK: %d\n",
+			ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget snd_proto_widget[] = {
+	SND_SOC_DAPM_MIC("Microphone Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route snd_proto_route[] = {
+	/* speaker connected to LHPOUT/RHPOUT */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Microphone Jack"},
+};
+
+/* audio machine driver */
+static struct snd_soc_card snd_proto = {
+	.name		= "snd_mikroe_proto",
+	.owner		= THIS_MODULE,
+	.dapm_widgets	= snd_proto_widget,
+	.num_dapm_widgets = ARRAY_SIZE(snd_proto_widget),
+	.dapm_routes	= snd_proto_route,
+	.num_dapm_routes = ARRAY_SIZE(snd_proto_route),
+};
+
+static int snd_proto_probe(struct platform_device *pdev)
+{
+	struct snd_soc_dai_link *dai;
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *codec_np, *cpu_np;
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	unsigned int dai_fmt;
+	int ret = 0;
+
+	if (!np) {
+		dev_err(&pdev->dev, "No device node supplied\n");
+		return -EINVAL;
+	}
+
+	snd_proto.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&snd_proto, "model");
+	if (ret)
+		return ret;
+
+	dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
+	snd_proto.dai_link = dai;
+	snd_proto.num_links = 1;
+
+	dai->name = "WM8731";
+	dai->stream_name = "WM8731 HiFi";
+	dai->codec_dai_name = "wm8731-hifi";
+	dai->init = &snd_proto_init;
+
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!codec_np) {
+		dev_err(&pdev->dev, "audio-codec node missing\n");
+		return -EINVAL;
+	}
+	dai->codec_of_node = codec_np;
+
+	cpu_np = of_parse_phandle(np, "i2s-controller", 0);
+	if (!cpu_np) {
+		dev_err(&pdev->dev, "i2s-controller missing\n");
+		return -EINVAL;
+	}
+	dai->cpu_of_node = cpu_np;
+	dai->platform_of_node = cpu_np;
+
+	dai_fmt = snd_soc_of_parse_daifmt(np, NULL,
+					  &bitclkmaster, &framemaster);
+	if (bitclkmaster != framemaster) {
+		dev_err(&pdev->dev, "Must be the same bitclock and frame master\n");
+		return -EINVAL;
+	}
+	if (bitclkmaster) {
+		dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+		if (codec_np == bitclkmaster)
+			dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+		else
+			dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	}
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+	dai->dai_fmt = dai_fmt;
+
+	of_node_put(codec_np);
+	of_node_put(cpu_np);
+
+	ret = snd_soc_register_card(&snd_proto);
+	if (ret && ret != -EPROBE_DEFER)
+		dev_err(&pdev->dev,
+			"snd_soc_register_card() failed: %d\n", ret);
+
+	return ret;
+}
+
+static int snd_proto_remove(struct platform_device *pdev)
+{
+	return snd_soc_unregister_card(&snd_proto);
+}
+
+static const struct of_device_id snd_proto_of_match[] = {
+	{ .compatible = "mikroe,mikroe-proto", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, snd_proto_of_match);
+
+static struct platform_driver snd_proto_driver = {
+	.driver = {
+		.name   = "snd-mikroe-proto",
+		.of_match_table = snd_proto_of_match,
+	},
+	.probe	  = snd_proto_probe,
+	.remove	 = snd_proto_remove,
+};
+
+module_platform_driver(snd_proto_driver);
+
+MODULE_AUTHOR("Florian Meier");
+MODULE_DESCRIPTION("ASoC Driver for PROTO board (WM8731)");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c
index 3a1393283156..214adcad5419 100644
--- a/sound/soc/atmel/tse850-pcm5142.c
+++ b/sound/soc/atmel/tse850-pcm5142.c
@@ -1,44 +1,38 @@
-/*
- * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
- *
- * Copyright (C) 2016 Axentia Technologies AB
- *
- * Author: Peter Rosin <peda@axentia.se>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-/*
- *               loop1 relays
- *   IN1 +---o  +------------+  o---+ OUT1
- *            \                /
- *             +              +
- *             |   /          |
- *             +--o  +--.     |
- *             |  add   |     |
- *             |        V     |
- *             |      .---.   |
- *   DAC +----------->|Sum|---+
- *             |      '---'   |
- *             |              |
- *             +              +
- *
- *   IN2 +---o--+------------+--o---+ OUT2
- *               loop2 relays
- *
- * The 'loop1' gpio pin controlls two relays, which are either in loop
- * position, meaning that input and output are directly connected, or
- * they are in mixer position, meaning that the signal is passed through
- * the 'Sum' mixer. Similarly for 'loop2'.
- *
- * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
- * mixer (if 'add' is active) and feeding the mixer output to OUT1. The
- * 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
- * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
- * of the (filtered) output from the PCM5142 codec.
- */
+// SPDX-License-Identifier: GPL-2.0
+//
+// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec
+//
+// Copyright (C) 2016 Axentia Technologies AB
+//
+// Author: Peter Rosin <peda@axentia.se>
+//
+//               loop1 relays
+//   IN1 +---o  +------------+  o---+ OUT1
+//            \                /
+//             +              +
+//             |   /          |
+//             +--o  +--.     |
+//             |  add   |     |
+//             |        V     |
+//             |      .---.   |
+//   DAC +----------->|Sum|---+
+//             |      '---'   |
+//             |              |
+//             +              +
+//
+//   IN2 +---o--+------------+--o---+ OUT2
+//               loop2 relays
+//
+// The 'loop1' gpio pin controlls two relays, which are either in loop
+// position, meaning that input and output are directly connected, or
+// they are in mixer position, meaning that the signal is passed through
+// the 'Sum' mixer. Similarly for 'loop2'.
+//
+// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the
+// mixer (if 'add' is active) and feeding the mixer output to OUT1. The
+// 'loop2' relays are active, short-cutting the TSE-850 from channel 2.
+// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name
+// of the (filtered) output from the PCM5142 codec.
 
 #include <linux/clk.h>
 #include <linux/gpio.h>
@@ -452,4 +446,4 @@ module_platform_driver(tse850_driver);
 /* Module information */
 MODULE_AUTHOR("Peter Rosin <peda@axentia.se>");
 MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index b733f1446353..b7c358b48d8d 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	cygaud->active_ports = 0;
 
 	dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
-	err = snd_soc_register_component(dev, &cygnus_ssp_component,
+	err = devm_snd_soc_register_component(dev, &cygnus_ssp_component,
 				cygnus_ssp_dai, active_port_count);
 	if (err) {
 		dev_err(dev, "snd_soc_register_dai failed\n");
@@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 	if (cygaud->irq_num <= 0) {
 		dev_err(dev, "platform_get_irq failed\n");
 		err = cygaud->irq_num;
-		goto err_irq;
+		return err;
 	}
 
 	err = audio_clk_init(pdev, cygaud);
 	if (err) {
 		dev_err(dev, "audio clock initialization failed\n");
-		goto err_irq;
+		return err;
 	}
 
 	err = cygnus_soc_platform_register(dev, cygaud);
 	if (err) {
 		dev_err(dev, "platform reg error %d\n", err);
-		goto err_irq;
+		return err;
 	}
 
 	return 0;
-
-err_irq:
-	snd_soc_unregister_component(dev);
-	return err;
 }
 
 static int cygnus_ssp_remove(struct platform_device *pdev)
 {
 	cygnus_soc_platform_unregister(&pdev->dev);
-	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index efb095dbcd71..9cc4f1848c9b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -82,6 +82,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ES7241
 	select SND_SOC_GTM601
 	select SND_SOC_HDAC_HDMI
+	select SND_SOC_HDAC_HDA
 	select SND_SOC_ICS43432
 	select SND_SOC_INNO_RK3036
 	select SND_SOC_ISABELLE if I2C
@@ -109,6 +110,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MT6351 if MTK_PMIC_WRAP
 	select SND_SOC_NAU8540 if I2C
 	select SND_SOC_NAU8810 if I2C
+	select SND_SOC_NAU8822 if I2C
 	select SND_SOC_NAU8824 if I2C
 	select SND_SOC_NAU8825 if I2C
 	select SND_SOC_HDMI_CODEC
@@ -119,6 +121,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_PCM186X_I2C if I2C
 	select SND_SOC_PCM186X_SPI if SPI_MASTER
 	select SND_SOC_PCM3008
+	select SND_SOC_PCM3060_I2C if I2C
+	select SND_SOC_PCM3060_SPI if SPI_MASTER
 	select SND_SOC_PCM3168A_I2C if I2C
 	select SND_SOC_PCM3168A_SPI if SPI_MASTER
 	select SND_SOC_PCM5102A
@@ -575,7 +579,11 @@ config SND_SOC_DA9055
 	tristate
 
 config SND_SOC_DMIC
-	tristate
+	tristate "Generic Digital Microphone CODEC"
+	depends on GPIOLIB
+	help
+	  Enable support for the Generic Digital Microphone CODEC.
+	  Select this if your sound card has DMICs.
 
 config SND_SOC_HDMI_CODEC
 	tristate
@@ -615,6 +623,10 @@ config SND_SOC_HDAC_HDMI
 	select SND_PCM_ELD
 	select HDMI
 
+config SND_SOC_HDAC_HDA
+	tristate
+	select SND_HDA
+
 config SND_SOC_ICS43432
 	tristate
 
@@ -629,7 +641,8 @@ config SND_SOC_LM49453
 	tristate
 
 config SND_SOC_MAX98088
-       tristate
+	tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
+	depends on I2C
 
 config SND_SOC_MAX98090
        tristate
@@ -732,6 +745,21 @@ config SND_SOC_PCM186X_SPI
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_PCM3060
+       tristate
+
+config SND_SOC_PCM3060_I2C
+	tristate "Texas Instruments PCM3060 CODEC - I2C"
+	depends on I2C
+	select SND_SOC_PCM3060
+	select REGMAP_I2C
+
+config SND_SOC_PCM3060_SPI
+	tristate "Texas Instruments PCM3060 CODEC - SPI"
+	depends on SPI_MASTER
+	select SND_SOC_PCM3060
+	select REGMAP_SPI
+
 config SND_SOC_PCM3168A
 	tristate
 
@@ -1299,6 +1327,10 @@ config SND_SOC_NAU8810
 	tristate "Nuvoton Technology Corporation NAU88C10 CODEC"
 	depends on I2C
 
+config SND_SOC_NAU8822
+	tristate "Nuvoton Technology Corporation NAU88C22 CODEC"
+	depends on I2C
+
 config SND_SOC_NAU8824
 	tristate "Nuvoton Technology Corporation NAU88L24 CODEC"
 	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 7ae7c85e8219..8ffab8c8dbfa 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -78,6 +78,7 @@ snd-soc-es8328-i2c-objs := es8328-i2c.o
 snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-gtm601-objs := gtm601.o
 snd-soc-hdac-hdmi-objs := hdac_hdmi.o
+snd-soc-hdac-hda-objs := hdac_hda.o
 snd-soc-ics43432-objs := ics43432.o
 snd-soc-inno-rk3036-objs := inno_rk3036.o
 snd-soc-isabelle-objs := isabelle.o
@@ -106,6 +107,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o
 snd-soc-mt6351-objs := mt6351.o
 snd-soc-nau8540-objs := nau8540.o
 snd-soc-nau8810-objs := nau8810.o
+snd-soc-nau8822-objs := nau8822.o
 snd-soc-nau8824-objs := nau8824.o
 snd-soc-nau8825-objs := nau8825.o
 snd-soc-hdmi-codec-objs := hdmi-codec.o
@@ -119,6 +121,9 @@ snd-soc-pcm186x-objs := pcm186x.o
 snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o
 snd-soc-pcm186x-spi-objs := pcm186x-spi.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm3060-objs := pcm3060.o
+snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o
+snd-soc-pcm3060-spi-objs := pcm3060-spi.o
 snd-soc-pcm3168a-objs := pcm3168a.o
 snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o
 snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o
@@ -338,6 +343,7 @@ obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
 obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_GTM601)    += snd-soc-gtm601.o
 obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o
+obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o
 obj-$(CONFIG_SND_SOC_ICS43432)	+= snd-soc-ics43432.o
 obj-$(CONFIG_SND_SOC_INNO_RK3036)	+= snd-soc-inno-rk3036.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
@@ -366,6 +372,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o
 obj-$(CONFIG_SND_SOC_MT6351)	+= snd-soc-mt6351.o
 obj-$(CONFIG_SND_SOC_NAU8540)   += snd-soc-nau8540.o
 obj-$(CONFIG_SND_SOC_NAU8810)   += snd-soc-nau8810.o
+obj-$(CONFIG_SND_SOC_NAU8822)   += snd-soc-nau8822.o
 obj-$(CONFIG_SND_SOC_NAU8824)   += snd-soc-nau8824.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
 obj-$(CONFIG_SND_SOC_HDMI_CODEC)	+= snd-soc-hdmi-codec.o
@@ -379,6 +386,9 @@ obj-$(CONFIG_SND_SOC_PCM186X)	+= snd-soc-pcm186x.o
 obj-$(CONFIG_SND_SOC_PCM186X_I2C)	+= snd-soc-pcm186x-i2c.o
 obj-$(CONFIG_SND_SOC_PCM186X_SPI)	+= snd-soc-pcm186x-spi.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM3060)	+= snd-soc-pcm3060.o
+obj-$(CONFIG_SND_SOC_PCM3060_I2C)	+= snd-soc-pcm3060-i2c.o
+obj-$(CONFIG_SND_SOC_PCM3060_SPI)	+= snd-soc-pcm3060-spi.o
 obj-$(CONFIG_SND_SOC_PCM3168A)	+= snd-soc-pcm3168a.o
 obj-$(CONFIG_SND_SOC_PCM3168A_I2C)	+= snd-soc-pcm3168a-i2c.o
 obj-$(CONFIG_SND_SOC_PCM3168A_SPI)	+= snd-soc-pcm3168a-spi.o
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index be136e981653..bef3e9e74c26 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component)
 			ARRAY_SIZE(adau1761_jack_detect_controls));
 		if (ret)
 			return ret;
-	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
+		/* fall through */
+	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE:
 		ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
 			ARRAY_SIZE(adau1761_no_dmic_routes));
 		if (ret)
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 57169b8ff14e..3959e6ad113d 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -21,11 +21,18 @@
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
 #include <linux/regmap.h>
+#include <asm/unaligned.h>
 
 #include "sigmadsp.h"
 #include "adau17x1.h"
 #include "adau-utils.h"
 
+#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006
+#define ADAU17X1_SAFELOAD_TRIGGER 0x0007
+#define ADAU17X1_SAFELOAD_DATA 0x0001
+#define ADAU17X1_SAFELOAD_DATA_SIZE 20
+#define ADAU17X1_WORD_SIZE 4
+
 static const char * const adau17x1_capture_mixer_boost_text[] = {
 	"Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3",
 };
@@ -60,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = {
 	SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum),
 };
 
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate);
+
 static int adau17x1_pll_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -313,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = {
 	{ "Capture", NULL, "Right Decimator" },
 };
 
-bool adau17x1_has_dsp(struct adau *adau)
+static bool adau17x1_has_dsp(struct adau *adau)
 {
 	switch (adau->type) {
 	case ADAU1761:
@@ -324,7 +334,17 @@ bool adau17x1_has_dsp(struct adau *adau)
 		return false;
 	}
 }
-EXPORT_SYMBOL_GPL(adau17x1_has_dsp);
+
+static bool adau17x1_has_safeload(struct adau *adau)
+{
+	switch (adau->type) {
+	case ADAU1761:
+	case ADAU1781:
+		return true;
+	default:
+		return false;
+	}
+}
 
 static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
 	int source, unsigned int freq_in, unsigned int freq_out)
@@ -836,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
+static int adau17x1_setup_firmware(struct snd_soc_component *component,
 	unsigned int rate)
 {
 	int ret;
@@ -880,7 +900,6 @@ err:
 
 	return ret;
 }
-EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
 int adau17x1_add_widgets(struct snd_soc_component *component)
 {
@@ -957,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component)
 }
 EXPORT_SYMBOL_GPL(adau17x1_resume);
 
+static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr,
+	const uint8_t bytes[], size_t len)
+{
+	uint8_t buf[ADAU17X1_WORD_SIZE];
+	uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE];
+	unsigned int addr_offset;
+	unsigned int nbr_words;
+	int ret;
+
+	/* write data to safeload addresses. Check if len is not a multiple of
+	 * 4 bytes, if so we need to zero pad.
+	 */
+	nbr_words = len / ADAU17X1_WORD_SIZE;
+	if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) {
+		ret = regmap_raw_write(sigmadsp->control_data,
+			ADAU17X1_SAFELOAD_DATA, bytes, len);
+	} else {
+		nbr_words++;
+		memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE);
+		memcpy(data, bytes, len);
+		ret = regmap_raw_write(sigmadsp->control_data,
+			ADAU17X1_SAFELOAD_DATA, data,
+			nbr_words * ADAU17X1_WORD_SIZE);
+	}
+
+	if (ret < 0)
+		return ret;
+
+	/* Write target address, target address is offset by 1 */
+	addr_offset = addr - 1;
+	put_unaligned_be32(addr_offset, buf);
+	ret = regmap_raw_write(sigmadsp->control_data,
+		ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE);
+	if (ret < 0)
+		return ret;
+
+	/* write nbr of words to trigger address */
+	put_unaligned_be32(nbr_words, buf);
+	ret = regmap_raw_write(sigmadsp->control_data,
+		ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static const struct sigmadsp_ops adau17x1_sigmadsp_ops = {
+	.safeload = adau17x1_safeload,
+};
+
 int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	enum adau17x1_type type, void (*switch_mode)(struct device *dev),
 	const char *firmware_name)
@@ -1002,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap,
 	dev_set_drvdata(dev, adau);
 
 	if (firmware_name) {
-		adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL,
-			firmware_name);
+		if (adau17x1_has_safeload(adau)) {
+			adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+				&adau17x1_sigmadsp_ops, firmware_name);
+		} else {
+			adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap,
+				NULL, firmware_name);
+		}
 		if (IS_ERR(adau->sigmadsp)) {
 			dev_warn(dev, "Could not find firmware file: %ld\n",
 				PTR_ERR(adau->sigmadsp));
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index e6fe87beec07..98a3b6f5bc96 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct snd_soc_component *component,
-	unsigned int rate);
-bool adau17x1_has_dsp(struct adau *adau);
-
 #define ADAU17X1_CLOCK_CONTROL			0x4000
 #define ADAU17X1_PLL_CONTROL			0x4002
 #define ADAU17X1_REC_POWER_MGMT			0x4009
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 407554175282..ab27d2b94d02 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -154,11 +154,11 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = {
 	SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
 				6, 1, 0),
 	SOC_ENUM("C Data Access", cam_mode_enum),
+	SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1),
 	SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
 				3, 1, 0),
 	SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
-	SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2,
-				0, 1, 0),
+	SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0),
 	SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
 	SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
 };
@@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = {
 	{"LINEOUTR", NULL, "DAC"},
 	{"SPDIFOUT", NULL, "SPDIF"},
 
+	{"Pre-amp MIC", NULL, "MICL"},
+	{"Pre-amp MIC", NULL, "MICR"},
+	{"ADC Mux", "MIC", "Pre-amp MIC"},
 	{"ADC Mux", "LINEIN", "LINEINL"},
 	{"ADC Mux", "LINEIN", "LINEINR"},
-	{"ADC Mux", "MIC", "MICL"},
-	{"ADC Mux", "MIC", "MICR"},
 	{"ADC", NULL, "ADC Mux"},
 	{"DOUT", NULL, "ADC"},
 	{"DAI1 Capture", NULL, "DOUT"},
@@ -496,7 +497,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component,
 			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
 
 #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
-			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
 
 static const struct snd_soc_dai_ops cs4265_ops = {
 	.hw_params	= cs4265_pcm_hw_params,
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 5080d7a3c279..fd2bd74024c1 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -21,6 +21,7 @@
  *  - master mode *NOT* supported
  */
 
+#include <linux/clk.h>
 #include <linux/module.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -41,6 +42,7 @@ enum master_slave_mode {
 
 struct cs42l51_private {
 	unsigned int mclk;
+	struct clk *mclk_handle;
 	unsigned int audio_mode;	/* The mode (I2S or left-justified) */
 	enum master_slave_mode func;
 };
@@ -237,6 +239,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
 		&cs42l51_adcr_mux_controls),
 };
 
+static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
+	SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+};
+
 static const struct snd_soc_dapm_route cs42l51_routes[] = {
 	{"HPL", NULL, "Left DAC"},
 	{"HPR", NULL, "Right DAC"},
@@ -487,6 +493,14 @@ static struct snd_soc_dai_driver cs42l51_dai = {
 static int cs42l51_component_probe(struct snd_soc_component *component)
 {
 	int ret, reg;
+	struct snd_soc_dapm_context *dapm;
+	struct cs42l51_private *cs42l51;
+
+	cs42l51 = snd_soc_component_get_drvdata(component);
+	dapm = snd_soc_component_get_dapm(component);
+
+	if (cs42l51->mclk_handle)
+		snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1);
 
 	/*
 	 * DAC configuration
@@ -540,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap)
 
 	dev_set_drvdata(dev, cs42l51);
 
+	cs42l51->mclk_handle = devm_clk_get(dev, "MCLK");
+	if (IS_ERR(cs42l51->mclk_handle)) {
+		if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT)
+			return PTR_ERR(cs42l51->mclk_handle);
+		cs42l51->mclk_handle = NULL;
+	}
+
 	/* Verify that we have a CS42L51 */
 	ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 8c4926df9286..71322e0410ee 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = {
 	{.compatible = "dmic-codec"},
 	{}
 };
+MODULE_DEVICE_TABLE(of, dmic_dev_match);
 
 static struct platform_driver dmic_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index e9fc2fd97d2f..3aedd609626c 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
 		break;
 	case 22579200:
 		mclkdiv2 = 1;
-		/* fallthru */
+		/* fall through */
 	case 11289600:
 		es8328->sysclk_constraints = &constraints_11289;
 		es8328->mclk_ratios = ratios_11289;
 		break;
 	case 24576000:
 		mclkdiv2 = 1;
-		/* fallthru */
+		/* fall through */
 	case 12288000:
 		es8328->sysclk_constraints = &constraints_12288;
 		es8328->mclk_ratios = ratios_12288;
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
new file mode 100644
index 000000000000..2aaa83028e55
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.c
@@ -0,0 +1,483 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * hdac_hda.c - ASoC extensions to reuse the legacy HDA codec drivers
+ * with ASoC platform drivers. These APIs are called by the legacy HDA
+ * codec drivers using hdac_ext_bus_ops ops.
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
+#include <sound/hda_register.h>
+#include "hdac_hda.h"
+
+#define HDAC_ANALOG_DAI_ID		0
+#define HDAC_DIGITAL_DAI_ID		1
+#define HDAC_ALT_ANALOG_DAI_ID		2
+
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+			SNDRV_PCM_FMTBIT_U8 | \
+			SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_U16_LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | \
+			SNDRV_PCM_FMTBIT_U24_LE | \
+			SNDRV_PCM_FMTBIT_S32_LE | \
+			SNDRV_PCM_FMTBIT_U32_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai);
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai);
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai);
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai);
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				     unsigned int tx_mask, unsigned int rx_mask,
+				     int slots, int slot_width);
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+						 struct snd_soc_dai *dai);
+
+static struct snd_soc_dai_ops hdac_hda_dai_ops = {
+	.startup = hdac_hda_dai_open,
+	.shutdown = hdac_hda_dai_close,
+	.prepare = hdac_hda_dai_prepare,
+	.hw_free = hdac_hda_dai_hw_free,
+	.set_tdm_slot = hdac_hda_dai_set_tdm_slot,
+};
+
+static struct snd_soc_dai_driver hdac_hda_dais[] = {
+{
+	.id = HDAC_ANALOG_DAI_ID,
+	.name = "Analog Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name	= "Analog Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 16,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= STUB_FORMATS,
+		.sig_bits	= 24,
+	},
+	.capture = {
+		.stream_name    = "Analog Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+},
+{
+	.id = HDAC_DIGITAL_DAI_ID,
+	.name = "Digital Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name    = "Digital Codec Playback",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates          = SNDRV_PCM_RATE_8000_192000,
+		.formats        = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+	.capture = {
+		.stream_name    = "Digital Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+},
+{
+	.id = HDAC_ALT_ANALOG_DAI_ID,
+	.name = "Alt Analog Codec DAI",
+	.ops = &hdac_hda_dai_ops,
+	.playback = {
+		.stream_name	= "Alt Analog Codec Playback",
+		.channels_min	= 1,
+		.channels_max	= 16,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= STUB_FORMATS,
+		.sig_bits	= 24,
+	},
+	.capture = {
+		.stream_name    = "Alt Analog Codec Capture",
+		.channels_min   = 1,
+		.channels_max   = 16,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = STUB_FORMATS,
+		.sig_bits = 24,
+	},
+}
+
+};
+
+static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai,
+				     unsigned int tx_mask, unsigned int rx_mask,
+				     int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hdac_hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = &hda_pvt->pcm[dai->id];
+	if (tx_mask)
+		pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+	else
+		pcm[dai->id].stream_tag[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+
+	return 0;
+}
+
+static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	hda_stream = &pcm->stream[substream->stream];
+	snd_hda_codec_cleanup(&hda_pvt->codec, hda_stream, substream);
+
+	return 0;
+}
+
+static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hdac_device *hdev;
+	struct hda_pcm_stream *hda_stream;
+	unsigned int format_val;
+	struct hda_pcm *pcm;
+	unsigned int stream;
+	int ret = 0;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	hdev = &hda_pvt->codec.core;
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	format_val = snd_hdac_calc_stream_format(runtime->rate,
+						 runtime->channels,
+						 runtime->format,
+						 hda_stream->maxbps,
+						 0);
+	if (!format_val) {
+		dev_err(&hdev->dev,
+			"invalid format_val, rate=%d, ch=%d, format=%d\n",
+			runtime->rate, runtime->channels, runtime->format);
+		return -EINVAL;
+	}
+
+	stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream];
+
+	ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream,
+				    stream, format_val, substream);
+	if (ret < 0)
+		dev_err(&hdev->dev, "codec prepare failed %d\n", ret);
+
+	return ret;
+}
+
+static int hdac_hda_dai_open(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+	int ret;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return -EINVAL;
+
+	snd_hda_codec_pcm_get(pcm);
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	ret = hda_stream->ops.open(hda_stream, &hda_pvt->codec, substream);
+	if (ret < 0)
+		snd_hda_codec_pcm_put(pcm);
+
+	return ret;
+}
+
+static void hdac_hda_dai_close(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct hdac_hda_priv *hda_pvt;
+	struct hda_pcm_stream *hda_stream;
+	struct hda_pcm *pcm;
+
+	hda_pvt = snd_soc_component_get_drvdata(component);
+	pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai);
+	if (!pcm)
+		return;
+
+	hda_stream = &pcm->stream[substream->stream];
+
+	hda_stream->ops.close(hda_stream, &hda_pvt->codec, substream);
+
+	snd_hda_codec_pcm_put(pcm);
+}
+
+static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt,
+						 struct snd_soc_dai *dai)
+{
+	struct hda_codec *hcodec = &hda_pvt->codec;
+	struct hda_pcm *cpcm;
+	const char *pcm_name;
+
+	switch (dai->id) {
+	case HDAC_ANALOG_DAI_ID:
+		pcm_name = "Analog";
+		break;
+	case HDAC_DIGITAL_DAI_ID:
+		pcm_name = "Digital";
+		break;
+	case HDAC_ALT_ANALOG_DAI_ID:
+		pcm_name = "Alt Analog";
+		break;
+	default:
+		dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id);
+		return NULL;
+	}
+
+	list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) {
+		if (strpbrk(cpcm->name, pcm_name))
+			return cpcm;
+	}
+
+	dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name);
+	return NULL;
+}
+
+static int hdac_hda_codec_probe(struct snd_soc_component *component)
+{
+	struct hdac_hda_priv *hda_pvt =
+			snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
+	struct hdac_device *hdev = &hda_pvt->codec.core;
+	struct hda_codec *hcodec = &hda_pvt->codec;
+	struct hdac_ext_link *hlink;
+	hda_codec_patch_t patch;
+	int ret;
+
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return -EIO;
+	}
+
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+	ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card,
+				       hdev->addr, hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "failed to create hda codec %d\n", ret);
+		goto error_no_pm;
+	}
+
+	/*
+	 * snd_hda_codec_device_new decrements the usage count so call get pm
+	 * else the device will be powered off
+	 */
+	pm_runtime_get_noresume(&hdev->dev);
+
+	hcodec->bus->card = dapm->card->snd_card;
+
+	ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name);
+		goto error;
+	}
+
+	ret = snd_hdac_regmap_init(&hcodec->core);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "regmap init failed\n");
+		goto error;
+	}
+
+	patch = (hda_codec_patch_t)hcodec->preset->driver_data;
+	if (patch) {
+		ret = patch(hcodec);
+		if (ret < 0) {
+			dev_err(&hdev->dev, "patch failed %d\n", ret);
+			goto error;
+		}
+	} else {
+		dev_dbg(&hdev->dev, "no patch file found\n");
+	}
+
+	ret = snd_hda_codec_parse_pcms(hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret);
+		goto error;
+	}
+
+	ret = snd_hda_codec_build_controls(hcodec);
+	if (ret < 0) {
+		dev_err(&hdev->dev, "unable to create controls %d\n", ret);
+		goto error;
+	}
+
+	hcodec->core.lazy_cache = true;
+
+	/*
+	 * hdac_device core already sets the state to active and calls
+	 * get_noresume. So enable runtime and set the device to suspend.
+	 * pm_runtime_enable is also called during codec registeration
+	 */
+	pm_runtime_put(&hdev->dev);
+	pm_runtime_suspend(&hdev->dev);
+
+	return 0;
+
+error:
+	pm_runtime_put(&hdev->dev);
+error_no_pm:
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+	return ret;
+}
+
+static void hdac_hda_codec_remove(struct snd_soc_component *component)
+{
+	struct hdac_hda_priv *hda_pvt =
+		      snd_soc_component_get_drvdata(component);
+	struct hdac_device *hdev = &hda_pvt->codec.core;
+	struct hdac_ext_link *hlink = NULL;
+
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return;
+	}
+
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+	pm_runtime_disable(&hdev->dev);
+}
+
+static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = {
+	{"AIF1TX", NULL, "Codec Input Pin1"},
+	{"AIF2TX", NULL, "Codec Input Pin2"},
+	{"AIF3TX", NULL, "Codec Input Pin3"},
+
+	{"Codec Output Pin1", NULL, "AIF1RX"},
+	{"Codec Output Pin2", NULL, "AIF2RX"},
+	{"Codec Output Pin3", NULL, "AIF3RX"},
+};
+
+static const struct snd_soc_dapm_widget hdac_hda_dapm_widgets[] = {
+	/* Audio Interface */
+	SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+
+	/* Input Pins */
+	SND_SOC_DAPM_INPUT("Codec Input Pin1"),
+	SND_SOC_DAPM_INPUT("Codec Input Pin2"),
+	SND_SOC_DAPM_INPUT("Codec Input Pin3"),
+
+	/* Output Pins */
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin1"),
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin2"),
+	SND_SOC_DAPM_OUTPUT("Codec Output Pin3"),
+};
+
+static const struct snd_soc_component_driver hdac_hda_codec = {
+	.probe		= hdac_hda_codec_probe,
+	.remove		= hdac_hda_codec_remove,
+	.idle_bias_on	= false,
+	.dapm_widgets           = hdac_hda_dapm_widgets,
+	.num_dapm_widgets       = ARRAY_SIZE(hdac_hda_dapm_widgets),
+	.dapm_routes            = hdac_hda_dapm_routes,
+	.num_dapm_routes        = ARRAY_SIZE(hdac_hda_dapm_routes),
+};
+
+static int hdac_hda_dev_probe(struct hdac_device *hdev)
+{
+	struct hdac_ext_link *hlink;
+	struct hdac_hda_priv *hda_pvt;
+	int ret;
+
+	/* hold the ref while we probe */
+	hlink = snd_hdac_ext_bus_get_link(hdev->bus, dev_name(&hdev->dev));
+	if (!hlink) {
+		dev_err(&hdev->dev, "hdac link not found\n");
+		return -EIO;
+	}
+	snd_hdac_ext_bus_link_get(hdev->bus, hlink);
+
+	hda_pvt = hdac_to_hda_priv(hdev);
+	if (!hda_pvt)
+		return -ENOMEM;
+
+	/* ASoC specific initialization */
+	ret = devm_snd_soc_register_component(&hdev->dev,
+					 &hdac_hda_codec, hdac_hda_dais,
+					 ARRAY_SIZE(hdac_hda_dais));
+	if (ret < 0) {
+		dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret);
+		return ret;
+	}
+
+	dev_set_drvdata(&hdev->dev, hda_pvt);
+	snd_hdac_ext_bus_link_put(hdev->bus, hlink);
+
+	return ret;
+}
+
+static int hdac_hda_dev_remove(struct hdac_device *hdev)
+{
+	return 0;
+}
+
+static struct hdac_ext_bus_ops hdac_ops = {
+	.hdev_attach = hdac_hda_dev_probe,
+	.hdev_detach = hdac_hda_dev_remove,
+};
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void)
+{
+	return &hdac_ops;
+}
+EXPORT_SYMBOL_GPL(snd_soc_hdac_hda_get_ops);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("ASoC Extensions for legacy HDA Drivers");
+MODULE_AUTHOR("Rakesh Ughreja<rakesh.a.ughreja@intel.com>");
diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h
new file mode 100644
index 000000000000..e444ef593360
--- /dev/null
+++ b/sound/soc/codecs/hdac_hda.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+#ifndef __HDAC_HDA_H__
+#define __HDAC_HDA_H__
+
+struct hdac_hda_pcm {
+	int stream_tag[2];
+};
+
+struct hdac_hda_priv {
+	struct hda_codec codec;
+	struct hdac_hda_pcm pcm[2];
+};
+
+#define hdac_to_hda_priv(_hdac) \
+			container_of(_hdac, struct hdac_hda_priv, codec.core)
+#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
+
+struct hdac_ext_bus_ops *snd_soc_hdac_hda_get_ops(void);
+
+#endif /* __HDAC_HDA_H__ */
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 7b8533abf637..4e9854889a95 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev,
 		if (ret)
 			return ret;
 
+		/* Filter out 44.1, 88.2 and 176.4Khz */
+		rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 |
+			   SNDRV_PCM_RATE_176400);
+		if (!rates)
+			return -EINVAL;
+
 		sprintf(dai_name, "intel-hdmi-hifi%d", i+1);
 		hdmi_dais[i].name = devm_kstrdup(&hdev->dev,
 					dai_name, GFP_KERNEL);
@@ -1598,7 +1604,7 @@ static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->pcm && (rtd->pcm->device == device))
 			return rtd->pcm;
 	}
@@ -1961,9 +1967,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx)
 
 	port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head);
 
-	if (!port)
-		return 0;
-
 	if (!port || !port->eld.eld_valid)
 		return 0;
 
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index fb515aaa54fc..ca172a4b6849 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -16,6 +16,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/regmap.h>
+#include <linux/clk.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -42,6 +43,7 @@ struct max98088_priv {
 	struct regmap *regmap;
 	enum max98088_type devtype;
 	struct max98088_pdata *pdata;
+	struct clk *mclk;
 	unsigned int sysclk;
 	struct max98088_cdata dai[2];
 	int eq_textcnt;
@@ -1103,6 +1105,11 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai,
        if (freq == max98088->sysclk)
                return 0;
 
+	if (!IS_ERR(max98088->mclk)) {
+		freq = clk_round_rate(max98088->mclk, freq);
+		clk_set_rate(max98088->mclk, freq);
+	}
+
        /* Setup clocks for slave mode, and using the PLL
         * PSCLK = 0x01 (when master clk is 10MHz to 20MHz)
         *         0x02 (when master clk is 20MHz to 30MHz)..
@@ -1310,6 +1317,20 @@ static int max98088_set_bias_level(struct snd_soc_component *component,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
+		/*
+		 * SND_SOC_BIAS_PREPARE is called while preparing for a
+		 * transition to ON or away from ON. If current bias_level
+		 * is SND_SOC_BIAS_ON, then it is preparing for a transition
+		 * away from ON. Disable the clock in that case, otherwise
+		 * enable it.
+		 */
+		if (!IS_ERR(max98088->mclk)) {
+			if (snd_soc_component_get_bias_level(component) ==
+			    SND_SOC_BIAS_ON)
+				clk_disable_unprepare(max98088->mclk);
+			else
+				clk_prepare_enable(max98088->mclk);
+		}
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1725,6 +1746,11 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        if (IS_ERR(max98088->regmap))
 	       return PTR_ERR(max98088->regmap);
 
+	max98088->mclk = devm_clk_get(&i2c->dev, "mclk");
+	if (IS_ERR(max98088->mclk))
+		if (PTR_ERR(max98088->mclk) == -EPROBE_DEFER)
+			return PTR_ERR(max98088->mclk);
+
        max98088->devtype = id->driver_data;
 
        i2c_set_clientdata(i2c, max98088);
@@ -1742,9 +1768,19 @@ static const struct i2c_device_id max98088_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id max98088_of_match[] = {
+	{ .compatible = "maxim,max98088" },
+	{ .compatible = "maxim,max98089" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, max98088_of_match);
+#endif
+
 static struct i2c_driver max98088_i2c_driver = {
 	.driver = {
 		.name = "max98088",
+		.of_match_table = of_match_ptr(max98088_of_match),
 	},
 	.probe  = max98088_i2c_probe,
 	.id_table = max98088_i2c_id,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 1093f766d0d2..a09d01318f79 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -2,6 +2,7 @@
 // Copyright (c) 2017, Maxim Integrated
 
 #include <linux/acpi.h>
+#include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/module.h>
 #include <linux/regmap.h>
@@ -454,7 +455,7 @@ SND_SOC_DAPM_SIGGEN("IMON"),
 SND_SOC_DAPM_SIGGEN("FBMON"),
 };
 
-static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0);
+static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1);
 static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv,
 	0, 8, TLV_DB_SCALE_ITEM(0, 50, 0),
 	9, 10, TLV_DB_SCALE_ITEM(500, 100, 0),
@@ -470,19 +471,19 @@ static const DECLARE_TLV_DB_RANGE(max98373_dht_spkgain_min_tlv,
 	0, 9, TLV_DB_SCALE_ITEM(800, 100, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_dht_rotation_point_tlv,
-	0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0),
-	2, 7, TLV_DB_SCALE_ITEM(-200, -100, 0),
-	8, 9, TLV_DB_SCALE_ITEM(-1000, -200, 0),
-	10, 11, TLV_DB_SCALE_ITEM(-1500, -300, 0),
-	12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0),
-	14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0),
+	0, 1, TLV_DB_SCALE_ITEM(-3000, 500, 0),
+	2, 4, TLV_DB_SCALE_ITEM(-2200, 200, 0),
+	5, 6, TLV_DB_SCALE_ITEM(-1500, 300, 0),
+	7, 9, TLV_DB_SCALE_ITEM(-1000, 200, 0),
+	10, 13, TLV_DB_SCALE_ITEM(-500, 100, 0),
+	14, 15, TLV_DB_SCALE_ITEM(-100, 50, 0),
 );
 static const DECLARE_TLV_DB_RANGE(max98373_limiter_thresh_tlv,
-	0, 15, TLV_DB_SCALE_ITEM(0, -100, 0),
+	0, 15, TLV_DB_SCALE_ITEM(-1500, 100, 0),
 );
 
 static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv,
-	0, 60, TLV_DB_SCALE_ITEM(0, -25, 0),
+	0, 60, TLV_DB_SCALE_ITEM(-1500, 25, 0),
 );
 
 static bool max98373_readable_register(struct device *dev, unsigned int reg)
@@ -604,7 +605,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG,
 SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG,
 	MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0),
 SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL,
-	0, 0x7F, 0, max98373_digital_tlv),
+	0, 0x7F, 1, max98373_digital_tlv),
 SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN,
 	MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv),
 SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN,
@@ -616,7 +617,7 @@ SOC_SINGLE("DHT Switch", MAX98373_R20D4_DHT_EN,
 SOC_SINGLE_TLV("DHT Min Volume", MAX98373_R20D1_DHT_CFG,
 	MAX98373_DHT_SPK_GAIN_MIN_SHIFT, 9, 0, max98373_dht_spkgain_min_tlv),
 SOC_SINGLE_TLV("DHT Rot Pnt Volume", MAX98373_R20D1_DHT_CFG,
-	MAX98373_DHT_ROT_PNT_SHIFT, 15, 0, max98373_dht_rotation_point_tlv),
+	MAX98373_DHT_ROT_PNT_SHIFT, 15, 1, max98373_dht_rotation_point_tlv),
 SOC_SINGLE_TLV("DHT Attack Step Volume", MAX98373_R20D2_DHT_ATTACK_CFG,
 	MAX98373_DHT_ATTACK_STEP_SHIFT, 4, 0, max98373_dht_step_size_tlv),
 SOC_SINGLE_TLV("DHT Release Step Volume", MAX98373_R20D3_DHT_RELEASE_CFG,
@@ -653,29 +654,29 @@ SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),
 SOC_SINGLE("BDE Attack Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 4, 0xF, 0),
 SOC_SINGLE("BDE Release Rate", MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0, 0xF, 0),
 SOC_SINGLE_TLV("BDE LVL1 Clip Thresh Volume", MAX98373_R20A9_BDE_L1_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Thresh Volume", MAX98373_R20AC_BDE_L2_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Thresh Volume", MAX98373_R20AF_BDE_L3_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Thresh Volume", MAX98373_R20B2_BDE_L4_CFG_2,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Clip Reduction Volume", MAX98373_R20AA_BDE_L1_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Clip Reduction Volume", MAX98373_R20AD_BDE_L2_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Clip Reduction Volume", MAX98373_R20B0_BDE_L3_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Clip Reduction Volume", MAX98373_R20B3_BDE_L4_CFG_3,
-	0, 0x3C, 0, max98373_bde_gain_tlv),
+	0, 0x3C, 1, max98373_bde_gain_tlv),
 SOC_SINGLE_TLV("BDE LVL1 Limiter Thresh Volume", MAX98373_R20A8_BDE_L1_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL2 Limiter Thresh Volume", MAX98373_R20AB_BDE_L2_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL3 Limiter Thresh Volume", MAX98373_R20AE_BDE_L3_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 SOC_SINGLE_TLV("BDE LVL4 Limiter Thresh Volume", MAX98373_R20B1_BDE_L4_CFG_1,
-	0, 0xF, 0, max98373_limiter_thresh_tlv),
+	0, 0xF, 1, max98373_limiter_thresh_tlv),
 /* Limiter */
 SOC_SINGLE("Limiter Switch", MAX98373_R20E2_LIMITER_EN,
 	MAX98373_LIMITER_EN_SHIFT, 1, 0),
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
new file mode 100644
index 000000000000..622ce947f134
--- /dev/null
+++ b/sound/soc/codecs/nau8822.c
@@ -0,0 +1,1136 @@
+/*
+ * nau8822.c  --  NAU8822 ALSA Soc Audio Codec driver
+ *
+ * Copyright 2017 Nuvoton Technology Corp.
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.com>
+ *
+ * Based on WM8974.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+#include "nau8822.h"
+
+#define NAU_PLL_FREQ_MAX 100000000
+#define NAU_PLL_FREQ_MIN 90000000
+#define NAU_PLL_REF_MAX 33000000
+#define NAU_PLL_REF_MIN 8000000
+#define NAU_PLL_OPTOP_MIN 6
+
+static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 };
+
+static const struct reg_default nau8822_reg_defaults[] = {
+	{ NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 },
+	{ NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 },
+	{ NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 },
+	{ NAU8822_REG_AUDIO_INTERFACE, 0x0050 },
+	{ NAU8822_REG_COMPANDING_CONTROL, 0x0000 },
+	{ NAU8822_REG_CLOCKING, 0x0140 },
+	{ NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 },
+	{ NAU8822_REG_GPIO_CONTROL, 0x0000 },
+	{ NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 },
+	{ NAU8822_REG_DAC_CONTROL, 0x0000 },
+	{ NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 },
+	{ NAU8822_REG_ADC_CONTROL, 0x0100 },
+	{ NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff },
+	{ NAU8822_REG_EQ1, 0x012c },
+	{ NAU8822_REG_EQ2, 0x002c },
+	{ NAU8822_REG_EQ3, 0x002c },
+	{ NAU8822_REG_EQ4, 0x002c },
+	{ NAU8822_REG_EQ5, 0x002c },
+	{ NAU8822_REG_DAC_LIMITER_1, 0x0032 },
+	{ NAU8822_REG_DAC_LIMITER_2, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_1, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_2, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_3, 0x0000 },
+	{ NAU8822_REG_NOTCH_FILTER_4, 0x0000 },
+	{ NAU8822_REG_ALC_CONTROL_1, 0x0038 },
+	{ NAU8822_REG_ALC_CONTROL_2, 0x000b },
+	{ NAU8822_REG_ALC_CONTROL_3, 0x0032 },
+	{ NAU8822_REG_NOISE_GATE, 0x0010 },
+	{ NAU8822_REG_PLL_N, 0x0008 },
+	{ NAU8822_REG_PLL_K1, 0x000c },
+	{ NAU8822_REG_PLL_K2, 0x0093 },
+	{ NAU8822_REG_PLL_K3, 0x00e9 },
+	{ NAU8822_REG_3D_CONTROL, 0x0000 },
+	{ NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 },
+	{ NAU8822_REG_INPUT_CONTROL, 0x0033 },
+	{ NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 },
+	{ NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 },
+	{ NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 },
+	{ NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 },
+	{ NAU8822_REG_OUTPUT_CONTROL, 0x0002 },
+	{ NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 },
+	{ NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 },
+	{ NAU8822_REG_LHP_VOLUME, 0x0039 },
+	{ NAU8822_REG_RHP_VOLUME, 0x0039 },
+	{ NAU8822_REG_LSPKOUT_VOLUME, 0x0039 },
+	{ NAU8822_REG_RSPKOUT_VOLUME, 0x0039 },
+	{ NAU8822_REG_AUX2_MIXER, 0x0001 },
+	{ NAU8822_REG_AUX1_MIXER, 0x0001 },
+	{ NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 },
+	{ NAU8822_REG_LEFT_TIME_SLOT, 0x0000 },
+	{ NAU8822_REG_MISC, 0x0020 },
+	{ NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 },
+	{ NAU8822_REG_DEVICE_REVISION, 0x007f },
+	{ NAU8822_REG_DEVICE_ID, 0x001a },
+	{ NAU8822_REG_DAC_DITHER, 0x0114 },
+	{ NAU8822_REG_ALC_ENHANCE_1, 0x0000 },
+	{ NAU8822_REG_ALC_ENHANCE_2, 0x0000 },
+	{ NAU8822_REG_192KHZ_SAMPLING, 0x0008 },
+	{ NAU8822_REG_MISC_CONTROL, 0x0000 },
+	{ NAU8822_REG_INPUT_TIEOFF, 0x0000 },
+	{ NAU8822_REG_POWER_REDUCTION, 0x0000 },
+	{ NAU8822_REG_AGC_PEAK2PEAK, 0x0000 },
+	{ NAU8822_REG_AGC_PEAK_DETECT, 0x0000 },
+	{ NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 },
+	{ NAU8822_REG_OUTPUT_TIEOFF, 0x0000 },
+};
+
+static bool nau8822_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+	case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+	case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+	case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+	case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+	case NAU8822_REG_3D_CONTROL:
+	case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+	case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+	case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+	case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+	case NAU8822_REG_DAC_DITHER:
+	case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+	case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool nau8822_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1:
+	case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME:
+	case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5:
+	case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2:
+	case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4:
+	case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3:
+	case NAU8822_REG_3D_CONTROL:
+	case NAU8822_REG_RIGHT_SPEAKER_CONTROL:
+	case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL:
+	case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER:
+	case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID:
+	case NAU8822_REG_DAC_DITHER:
+	case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL:
+	case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool nau8822_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case NAU8822_REG_RESET:
+	case NAU8822_REG_DEVICE_REVISION:
+	case NAU8822_REG_DEVICE_ID:
+	case NAU8822_REG_AGC_PEAK2PEAK:
+	case NAU8822_REG_AGC_PEAK_DETECT:
+	case NAU8822_REG_AUTOMUTE_CONTROL:
+		return true;
+	default:
+		return false;
+	}
+}
+
+/* The EQ parameters get function is to get the 5 band equalizer control.
+ * The regmap raw read can't work here because regmap doesn't provide
+ * value format for value width of 9 bits. Therefore, the driver reads data
+ * from cache and makes value format according to the endianness of
+ * bytes type control element.
+ */
+static int nau8822_eq_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+			snd_soc_kcontrol_component(kcontrol);
+	struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+	int i, reg;
+	u16 reg_val, *val;
+
+	val = (u16 *)ucontrol->value.bytes.data;
+	reg = NAU8822_REG_EQ1;
+	for (i = 0; i < params->max / sizeof(u16); i++) {
+		reg_val = snd_soc_component_read32(component, reg + i);
+		/* conversion of 16-bit integers between native CPU format
+		 * and big endian format
+		 */
+		reg_val = cpu_to_be16(reg_val);
+		memcpy(val + i, &reg_val, sizeof(reg_val));
+	}
+
+	return 0;
+}
+
+/* The EQ parameters put function is to make configuration of 5 band equalizer
+ * control. These configuration includes central frequency, equalizer gain,
+ * cut-off frequency, bandwidth control, and equalizer path.
+ * The regmap raw write can't work here because regmap doesn't provide
+ * register and value format for register with address 7 bits and value 9 bits.
+ * Therefore, the driver makes value format according to the endianness of
+ * bytes type control element and writes data to codec.
+ */
+static int nau8822_eq_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+			snd_soc_kcontrol_component(kcontrol);
+	struct soc_bytes_ext *params = (void *)kcontrol->private_value;
+	void *data;
+	u16 *val, value;
+	int i, reg, ret;
+
+	data = kmemdup(ucontrol->value.bytes.data,
+		params->max, GFP_KERNEL | GFP_DMA);
+	if (!data)
+		return -ENOMEM;
+
+	val = (u16 *)data;
+	reg = NAU8822_REG_EQ1;
+	for (i = 0; i < params->max / sizeof(u16); i++) {
+		/* conversion of 16-bit integers between native CPU format
+		 * and big endian format
+		 */
+		value = be16_to_cpu(*(val + i));
+		ret = snd_soc_component_write(component, reg + i, value);
+		if (ret) {
+			dev_err(component->dev,
+			    "EQ configuration fail, register: %x ret: %d\n",
+			    reg + i, ret);
+			kfree(data);
+			return ret;
+		}
+	}
+	kfree(data);
+
+	return 0;
+}
+
+static const char * const nau8822_companding[] = {
+	"Off", "NC", "u-law", "A-law"};
+
+static const struct soc_enum nau8822_companding_adc_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT,
+		ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const struct soc_enum nau8822_companding_dac_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT,
+		ARRAY_SIZE(nau8822_companding), nau8822_companding);
+
+static const char * const nau8822_eqmode[] = {"Capture", "Playback"};
+
+static const struct soc_enum nau8822_eqmode_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT,
+		ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode);
+
+static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"};
+static const char * const nau8822_alc3[] = {"Normal", "Limiter"};
+
+static const struct soc_enum nau8822_alc_enable_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT,
+		ARRAY_SIZE(nau8822_alc1), nau8822_alc1);
+
+static const struct soc_enum nau8822_alc_mode_enum =
+	SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT,
+		ARRAY_SIZE(nau8822_alc3), nau8822_alc3);
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
+
+static const struct snd_kcontrol_new nau8822_snd_controls[] = {
+	SOC_ENUM("ADC Companding", nau8822_companding_adc_enum),
+	SOC_ENUM("DAC Companding", nau8822_companding_dac_enum),
+
+	SOC_ENUM("EQ Function", nau8822_eqmode_enum),
+	SND_SOC_BYTES_EXT("EQ Parameters", 10,
+		  nau8822_eq_get, nau8822_eq_put),
+
+	SOC_DOUBLE("DAC Inversion Switch",
+		NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0),
+	SOC_DOUBLE_R_TLV("PCM Volume",
+		NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+		NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+	SOC_SINGLE("High Pass Filter Switch",
+		NAU8822_REG_ADC_CONTROL, 8, 1, 0),
+	SOC_SINGLE("High Pass Cut Off",
+		NAU8822_REG_ADC_CONTROL, 4, 7, 0),
+
+	SOC_DOUBLE("ADC Inversion Switch",
+		NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0),
+	SOC_DOUBLE_R_TLV("ADC Volume",
+		NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+		NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv),
+
+	SOC_SINGLE("DAC Limiter Switch",
+		NAU8822_REG_DAC_LIMITER_1, 8, 1, 0),
+	SOC_SINGLE("DAC Limiter Decay",
+		NAU8822_REG_DAC_LIMITER_1, 4, 15, 0),
+	SOC_SINGLE("DAC Limiter Attack",
+		NAU8822_REG_DAC_LIMITER_1, 0, 15, 0),
+	SOC_SINGLE("DAC Limiter Threshold",
+		NAU8822_REG_DAC_LIMITER_2, 4, 7, 0),
+	SOC_SINGLE_TLV("DAC Limiter Volume",
+		NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
+
+	SOC_ENUM("ALC Mode", nau8822_alc_mode_enum),
+	SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum),
+	SOC_SINGLE("ALC Min Gain",
+		NAU8822_REG_ALC_CONTROL_1, 0, 7, 0),
+	SOC_SINGLE("ALC Max Gain",
+		NAU8822_REG_ALC_CONTROL_1, 3, 7, 0),
+	SOC_SINGLE("ALC Hold",
+		NAU8822_REG_ALC_CONTROL_2, 4, 10, 0),
+	SOC_SINGLE("ALC Target",
+		NAU8822_REG_ALC_CONTROL_2, 0, 15, 0),
+	SOC_SINGLE("ALC Decay",
+		NAU8822_REG_ALC_CONTROL_3, 4, 10, 0),
+	SOC_SINGLE("ALC Attack",
+		NAU8822_REG_ALC_CONTROL_3, 0, 10, 0),
+	SOC_SINGLE("ALC Noise Gate Switch",
+		NAU8822_REG_NOISE_GATE, 3, 1, 0),
+	SOC_SINGLE("ALC Noise Gate Threshold",
+		NAU8822_REG_NOISE_GATE, 0, 7, 0),
+
+	SOC_DOUBLE_R("PGA ZC Switch",
+		NAU8822_REG_LEFT_INP_PGA_CONTROL,
+		NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+		7, 1, 0),
+	SOC_DOUBLE_R_TLV("PGA Volume",
+		NAU8822_REG_LEFT_INP_PGA_CONTROL,
+		NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv),
+
+	SOC_DOUBLE_R("Headphone ZC Switch",
+		NAU8822_REG_LHP_VOLUME,
+		NAU8822_REG_RHP_VOLUME, 7, 1, 0),
+	SOC_DOUBLE_R("Headphone Playback Switch",
+		NAU8822_REG_LHP_VOLUME,
+		NAU8822_REG_RHP_VOLUME, 6, 1, 1),
+	SOC_DOUBLE_R_TLV("Headphone Volume",
+		NAU8822_REG_LHP_VOLUME,
+		NAU8822_REG_RHP_VOLUME,	0, 63, 0, spk_tlv),
+
+	SOC_DOUBLE_R("Speaker ZC Switch",
+		NAU8822_REG_LSPKOUT_VOLUME,
+		NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0),
+	SOC_DOUBLE_R("Speaker Playback Switch",
+		NAU8822_REG_LSPKOUT_VOLUME,
+		NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1),
+	SOC_DOUBLE_R_TLV("Speaker Volume",
+		NAU8822_REG_LSPKOUT_VOLUME,
+		NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv),
+
+	SOC_DOUBLE_R("AUXOUT Playback Switch",
+		NAU8822_REG_AUX2_MIXER,
+		NAU8822_REG_AUX1_MIXER, 6, 1, 1),
+
+	SOC_DOUBLE_R_TLV("PGA Boost Volume",
+		NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+		NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv),
+	SOC_DOUBLE_R_TLV("L2/R2 Boost Volume",
+		NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+		NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv),
+	SOC_DOUBLE_R_TLV("Aux Boost Volume",
+		NAU8822_REG_LEFT_ADC_BOOST_CONTROL,
+		NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv),
+
+	SOC_SINGLE("DAC 128x Oversampling Switch",
+		NAU8822_REG_DAC_CONTROL, 5, 1, 0),
+	SOC_SINGLE("ADC 128x Oversampling Switch",
+		NAU8822_REG_ADC_CONTROL, 5, 1, 0),
+};
+
+/* LMAIN and RMAIN Mixer */
+static const struct snd_kcontrol_new nau8822_left_out_mixer[] = {
+	SOC_DAPM_SINGLE("LINMIX Switch",
+		NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("LAUX Switch",
+		NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0),
+	SOC_DAPM_SINGLE("LDAC Switch",
+		NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("RDAC Switch",
+		NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_right_out_mixer[] = {
+	SOC_DAPM_SINGLE("RINMIX Switch",
+		NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("RAUX Switch",
+		NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0),
+	SOC_DAPM_SINGLE("RDAC Switch",
+		NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("LDAC Switch",
+		NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0),
+};
+
+/* AUX1 and AUX2 Mixer */
+static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = {
+	SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0),
+	SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0),
+	SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = {
+	SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0),
+	SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("AUX1MIX Output Switch",
+		NAU8822_REG_AUX2_MIXER, 3, 1, 0),
+};
+
+/* Input PGA */
+static const struct snd_kcontrol_new nau8822_left_input_mixer[] = {
+	SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0),
+};
+static const struct snd_kcontrol_new nau8822_right_input_mixer[] = {
+	SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0),
+	SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0),
+	SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0),
+};
+
+/* Loopback Switch */
+static const struct snd_kcontrol_new nau8822_loopback =
+	SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL,
+		NAU8822_ADDAP_SFT, 1, 0);
+
+static int check_mclk_select_pll(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(source->dapm);
+	unsigned int value;
+
+	value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING);
+
+	return (value & NAU8822_CLKM_MASK);
+}
+
+static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+		NAU8822_REG_POWER_MANAGEMENT_3, 0, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+		NAU8822_REG_POWER_MANAGEMENT_3, 1, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
+		NAU8822_REG_POWER_MANAGEMENT_2, 0, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+		NAU8822_REG_POWER_MANAGEMENT_2, 1, 0),
+
+	SOC_MIXER_ARRAY("Left Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer),
+	SOC_MIXER_ARRAY("Right Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_3,	3, 0, nau8822_right_out_mixer),
+	SOC_MIXER_ARRAY("AUX1 Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer),
+	SOC_MIXER_ARRAY("AUX2 Output Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_1,	6, 0, nau8822_auxout2_mixer),
+
+	SOC_MIXER_ARRAY("Left Input Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2,
+		2, 0, nau8822_left_input_mixer),
+	SOC_MIXER_ARRAY("Right Input Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2,
+		3, 0, nau8822_right_input_mixer),
+
+	SND_SOC_DAPM_PGA("Left Boost Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Boost Mixer",
+		NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Left Capture PGA",
+		NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Capture PGA",
+		NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Left Headphone Out",
+		NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Headphone Out",
+		NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Left Speaker Out",
+		 NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Speaker Out",
+		NAU8822_REG_POWER_MANAGEMENT_3,	5, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("AUX1 Out",
+		NAU8822_REG_POWER_MANAGEMENT_3,	8, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX2 Out",
+		NAU8822_REG_POWER_MANAGEMENT_3,	7, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("Mic Bias",
+		NAU8822_REG_POWER_MANAGEMENT_1,	4, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL",
+		NAU8822_REG_POWER_MANAGEMENT_1,	5, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0,
+		&nau8822_loopback),
+
+	SND_SOC_DAPM_INPUT("LMICN"),
+	SND_SOC_DAPM_INPUT("LMICP"),
+	SND_SOC_DAPM_INPUT("RMICN"),
+	SND_SOC_DAPM_INPUT("RMICP"),
+	SND_SOC_DAPM_INPUT("LAUX"),
+	SND_SOC_DAPM_INPUT("RAUX"),
+	SND_SOC_DAPM_INPUT("L2"),
+	SND_SOC_DAPM_INPUT("R2"),
+	SND_SOC_DAPM_OUTPUT("LHP"),
+	SND_SOC_DAPM_OUTPUT("RHP"),
+	SND_SOC_DAPM_OUTPUT("LSPK"),
+	SND_SOC_DAPM_OUTPUT("RSPK"),
+	SND_SOC_DAPM_OUTPUT("AUXOUT1"),
+	SND_SOC_DAPM_OUTPUT("AUXOUT2"),
+};
+
+static const struct snd_soc_dapm_route nau8822_dapm_routes[] = {
+	{"Right DAC", NULL, "PLL", check_mclk_select_pll},
+	{"Left DAC", NULL, "PLL", check_mclk_select_pll},
+
+	/* LMAIN and RMAIN Mixer */
+	{"Right Output Mixer", "LDAC Switch", "Left DAC"},
+	{"Right Output Mixer", "RDAC Switch", "Right DAC"},
+	{"Right Output Mixer", "RAUX Switch", "RAUX"},
+	{"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+
+	{"Left Output Mixer", "LDAC Switch", "Left DAC"},
+	{"Left Output Mixer", "RDAC Switch", "Right DAC"},
+	{"Left Output Mixer", "LAUX Switch", "LAUX"},
+	{"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+
+	/* AUX1 and AUX2 Mixer */
+	{"AUX1 Output Mixer", "RDAC Switch", "Right DAC"},
+	{"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"},
+	{"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"},
+	{"AUX1 Output Mixer", "LDAC Switch", "Left DAC"},
+	{"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+
+	{"AUX2 Output Mixer", "LDAC Switch", "Left DAC"},
+	{"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"},
+	{"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"},
+	{"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"},
+
+	/* Outputs */
+	{"Right Headphone Out", NULL, "Right Output Mixer"},
+	{"RHP", NULL, "Right Headphone Out"},
+
+	{"Left Headphone Out", NULL, "Left Output Mixer"},
+	{"LHP", NULL, "Left Headphone Out"},
+
+	{"Right Speaker Out", NULL, "Right Output Mixer"},
+	{"RSPK", NULL, "Right Speaker Out"},
+
+	{"Left Speaker Out", NULL, "Left Output Mixer"},
+	{"LSPK", NULL, "Left Speaker Out"},
+
+	{"AUX1 Out", NULL, "AUX1 Output Mixer"},
+	{"AUX2 Out", NULL, "AUX2 Output Mixer"},
+	{"AUXOUT1", NULL, "AUX1 Out"},
+	{"AUXOUT2", NULL, "AUX2 Out"},
+
+	/* Boost Mixer */
+	{"Right ADC", NULL, "PLL", check_mclk_select_pll},
+	{"Left ADC", NULL, "PLL", check_mclk_select_pll},
+
+	{"Right ADC", NULL, "Right Boost Mixer"},
+
+	{"Right Boost Mixer", NULL, "RAUX"},
+	{"Right Boost Mixer", NULL, "Right Capture PGA"},
+	{"Right Boost Mixer", NULL, "R2"},
+
+	{"Left ADC", NULL, "Left Boost Mixer"},
+
+	{"Left Boost Mixer", NULL, "LAUX"},
+	{"Left Boost Mixer", NULL, "Left Capture PGA"},
+	{"Left Boost Mixer", NULL, "L2"},
+
+	/* Input PGA */
+	{"Right Capture PGA", NULL, "Right Input Mixer"},
+	{"Left Capture PGA", NULL, "Left Input Mixer"},
+
+	/* Enable Microphone Power */
+	{"Right Capture PGA", NULL, "Mic Bias"},
+	{"Left Capture PGA", NULL, "Mic Bias"},
+
+	{"Right Input Mixer", "R2 Switch", "R2"},
+	{"Right Input Mixer", "MicN Switch", "RMICN"},
+	{"Right Input Mixer", "MicP Switch", "RMICP"},
+
+	{"Left Input Mixer", "L2 Switch", "L2"},
+	{"Left Input Mixer", "MicN Switch", "LMICN"},
+	{"Left Input Mixer", "MicP Switch", "LMICP"},
+
+	/* Digital Loopback */
+	{"Digital Loopback", "Switch", "Left ADC"},
+	{"Digital Loopback", "Switch", "Right ADC"},
+	{"Left DAC", NULL, "Digital Loopback"},
+	{"Right DAC", NULL, "Digital Loopback"},
+};
+
+static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				 unsigned int freq, int dir)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+	nau8822->div_id = clk_id;
+	nau8822->sysclk = freq;
+	dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq,
+		clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK");
+
+	return 0;
+}
+
+static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs,
+				struct nau8822_pll *pll_param)
+{
+	u64 f2, f2_max, pll_ratio;
+	int i, scal_sel;
+
+	if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN)
+		return -EINVAL;
+	f2_max = 0;
+	scal_sel = ARRAY_SIZE(nau8822_mclk_scaler);
+
+	for (i = 0; i < scal_sel; i++) {
+		f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10;
+		if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX &&
+			f2_max < f2) {
+			f2_max = f2;
+			scal_sel = i;
+		}
+	}
+
+	if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel)
+		return -EINVAL;
+	pll_param->mclk_scaler = scal_sel;
+	f2 = f2_max;
+
+	/* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional
+	 * input; round up the 24+4bit.
+	 */
+	pll_ratio = div_u64(f2 << 28, pll_in);
+	pll_param->pre_factor = 0;
+	if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) {
+		pll_ratio <<= 1;
+		pll_param->pre_factor = 1;
+	}
+	pll_param->pll_int = (pll_ratio >> 28) & 0xF;
+	pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4);
+
+	return 0;
+}
+
+static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+	struct nau8822_pll *pll = &nau8822->pll;
+	int i, sclk, imclk;
+
+	switch (nau8822->div_id) {
+	case NAU8822_CLK_MCLK:
+		/* Configure the master clock prescaler div to make system
+		 * clock to approximate the internal master clock (IMCLK);
+		 * and large or equal to IMCLK.
+		 */
+		div = 0;
+		imclk = rate * 256;
+		for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) {
+			sclk = (nau8822->sysclk * 10) /	nau8822_mclk_scaler[i];
+			if (sclk < imclk)
+				break;
+			div = i;
+		}
+		dev_dbg(component->dev, "master clock prescaler %x for fs %d\n",
+			div, rate);
+
+		/* master clock from MCLK and disable PLL */
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+			(div << NAU8822_MCLKSEL_SFT));
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+			NAU8822_CLKM_MCLK);
+		break;
+
+	case NAU8822_CLK_PLL:
+		/* master clock from PLL and enable PLL */
+		if (pll->mclk_scaler != div) {
+			dev_err(component->dev,
+			"master clock prescaler not meet PLL parameters\n");
+			return -EINVAL;
+		}
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+			(div << NAU8822_MCLKSEL_SFT));
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK,
+			NAU8822_CLKM_PLL);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+				unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+	struct nau8822_pll *pll_param = &nau8822->pll;
+	int ret, fs;
+
+	fs = freq_out / 256;
+
+	ret = nau8822_calc_pll(freq_in, fs, pll_param);
+	if (ret < 0) {
+		dev_err(component->dev, "Unsupported input clock %d\n",
+			freq_in);
+		return ret;
+	}
+
+	dev_info(component->dev,
+		"pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n",
+		pll_param->pll_int, pll_param->pll_frac,
+		pll_param->mclk_scaler, pll_param->pre_factor);
+
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK,
+		(pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) |
+		pll_param->pll_int);
+	snd_soc_component_write(component,
+		NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) &
+		NAU8822_PLLK1_MASK);
+	snd_soc_component_write(component,
+		NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) &
+		NAU8822_PLLK2_MASK);
+	snd_soc_component_write(component,
+		NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK,
+		pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL);
+
+	return 0;
+}
+
+static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *component = dai->component;
+	u16 ctrl1_val = 0, ctrl2_val = 0;
+
+	dev_dbg(component->dev, "%s\n", __func__);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		ctrl2_val |= 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ctrl2_val &= ~1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ctrl1_val |= 0x10;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		ctrl1_val |= 0x8;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		ctrl1_val |= 0x18;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		ctrl1_val |= 0x180;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		ctrl1_val |= 0x100;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		ctrl1_val |= 0x80;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_AUDIO_INTERFACE,
+		NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK,
+		ctrl1_val);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val);
+
+	return 0;
+}
+
+static int nau8822_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+	int val_len = 0, val_rate = 0;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val_len |= NAU8822_WLEN_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val_len |= NAU8822_WLEN_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		val_len |= NAU8822_WLEN_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 8000:
+		val_rate |= NAU8822_SMPLR_8K;
+		break;
+	case 11025:
+		val_rate |= NAU8822_SMPLR_12K;
+		break;
+	case 16000:
+		val_rate |= NAU8822_SMPLR_16K;
+		break;
+	case 22050:
+		val_rate |= NAU8822_SMPLR_24K;
+		break;
+	case 32000:
+		val_rate |= NAU8822_SMPLR_32K;
+		break;
+	case 44100:
+	case 48000:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len);
+	snd_soc_component_update_bits(component,
+		NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate);
+
+	/* If the master clock is from MCLK, provide the runtime FS for driver
+	 * to get the master clock prescaler configuration.
+	 */
+	if (nau8822->div_id == NAU8822_CLK_MCLK)
+		nau8822_config_clkdiv(dai, 0, params_rate(params));
+
+	return 0;
+}
+
+static int nau8822_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_component *component = dai->component;
+
+	dev_dbg(component->dev, "%s: %d\n", __func__, mute);
+
+	if (mute)
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_DAC_CONTROL, 0x40, 0x40);
+	else
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_DAC_CONTROL, 0x40, 0);
+
+	return 0;
+}
+
+static int nau8822_set_bias_level(struct snd_soc_component *component,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_POWER_MANAGEMENT_1,
+			NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_POWER_MANAGEMENT_1,
+			NAU8822_IOBUF_EN | NAU8822_ABIAS_EN,
+			NAU8822_IOBUF_EN | NAU8822_ABIAS_EN);
+
+		if (snd_soc_component_get_bias_level(component) ==
+			SND_SOC_BIAS_OFF) {
+			snd_soc_component_update_bits(component,
+				NAU8822_REG_POWER_MANAGEMENT_1,
+				NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K);
+			mdelay(100);
+		}
+		snd_soc_component_update_bits(component,
+			NAU8822_REG_POWER_MANAGEMENT_1,
+			NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_component_write(component,
+			NAU8822_REG_POWER_MANAGEMENT_1, 0);
+		snd_soc_component_write(component,
+			NAU8822_REG_POWER_MANAGEMENT_2, 0);
+		snd_soc_component_write(component,
+			NAU8822_REG_POWER_MANAGEMENT_3, 0);
+		break;
+	}
+
+	dev_dbg(component->dev, "%s: %d\n", __func__, level);
+
+	return 0;
+}
+
+#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000)
+
+#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops nau8822_dai_ops = {
+	.hw_params	= nau8822_hw_params,
+	.digital_mute	= nau8822_mute,
+	.set_fmt	= nau8822_set_dai_fmt,
+	.set_sysclk	= nau8822_set_dai_sysclk,
+	.set_pll	= nau8822_set_pll,
+};
+
+static struct snd_soc_dai_driver nau8822_dai = {
+	.name = "nau8822-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = NAU8822_RATES,
+		.formats = NAU8822_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = NAU8822_RATES,
+		.formats = NAU8822_FORMATS,
+	},
+	.ops = &nau8822_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static int nau8822_suspend(struct snd_soc_component *component)
+{
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+	snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
+
+	regcache_mark_dirty(nau8822->regmap);
+
+	return 0;
+}
+
+static int nau8822_resume(struct snd_soc_component *component)
+{
+	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
+
+	regcache_sync(nau8822->regmap);
+
+	snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+/*
+ * These registers contain an "update" bit - bit 8. This means, for example,
+ * that one can write new DAC digital volume for both channels, but only when
+ * the update bit is set, will also the volume be updated - simultaneously for
+ * both channels.
+ */
+static const int update_reg[] = {
+	NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME,
+	NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME,
+	NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME,
+	NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME,
+	NAU8822_REG_LEFT_INP_PGA_CONTROL,
+	NAU8822_REG_RIGHT_INP_PGA_CONTROL,
+	NAU8822_REG_LHP_VOLUME,
+	NAU8822_REG_RHP_VOLUME,
+	NAU8822_REG_LSPKOUT_VOLUME,
+	NAU8822_REG_RSPKOUT_VOLUME,
+};
+
+static int nau8822_probe(struct snd_soc_component *component)
+{
+	int i;
+
+	/*
+	 * Set the update bit in all registers, that have one. This way all
+	 * writes to those registers will also cause the update bit to be
+	 * written.
+	 */
+	for (i = 0; i < ARRAY_SIZE(update_reg); i++)
+		snd_soc_component_update_bits(component,
+			update_reg[i], 0x100, 0x100);
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver soc_component_dev_nau8822 = {
+	.probe				= nau8822_probe,
+	.suspend			= nau8822_suspend,
+	.resume				= nau8822_resume,
+	.set_bias_level			= nau8822_set_bias_level,
+	.controls			= nau8822_snd_controls,
+	.num_controls			= ARRAY_SIZE(nau8822_snd_controls),
+	.dapm_widgets			= nau8822_dapm_widgets,
+	.num_dapm_widgets		= ARRAY_SIZE(nau8822_dapm_widgets),
+	.dapm_routes			= nau8822_dapm_routes,
+	.num_dapm_routes		= ARRAY_SIZE(nau8822_dapm_routes),
+	.idle_bias_on			= 1,
+	.use_pmdown_time		= 1,
+	.endianness			= 1,
+	.non_legacy_dai_naming		= 1,
+};
+
+static const struct regmap_config nau8822_regmap_config = {
+	.reg_bits = 7,
+	.val_bits = 9,
+
+	.max_register = NAU8822_REG_MAX_REGISTER,
+	.volatile_reg = nau8822_volatile,
+
+	.readable_reg = nau8822_readable_reg,
+	.writeable_reg = nau8822_writeable_reg,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = nau8822_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults),
+};
+
+static int nau8822_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct device *dev = &i2c->dev;
+	struct nau8822 *nau8822 = dev_get_platdata(dev);
+	int ret;
+
+	if (!nau8822) {
+		nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL);
+		if (nau8822 == NULL)
+			return -ENOMEM;
+	}
+	i2c_set_clientdata(i2c, nau8822);
+
+	nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config);
+	if (IS_ERR(nau8822->regmap)) {
+		ret = PTR_ERR(nau8822->regmap);
+		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+		return ret;
+	}
+	nau8822->dev = dev;
+
+	/* Reset the codec */
+	ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822,
+						&nau8822_dai, 1);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct i2c_device_id nau8822_i2c_id[] = {
+	{ "nau8822", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id nau8822_of_match[] = {
+	{ .compatible = "nuvoton,nau8822", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, nau8822_of_match);
+#endif
+
+static struct i2c_driver nau8822_i2c_driver = {
+	.driver = {
+		.name = "nau8822",
+		.of_match_table = of_match_ptr(nau8822_of_match),
+	},
+	.probe =    nau8822_i2c_probe,
+	.id_table = nau8822_i2c_id,
+};
+module_i2c_driver(nau8822_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC NAU8822 codec driver");
+MODULE_AUTHOR("David Lin <ctlin0@nuvoton.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h
new file mode 100644
index 000000000000..aa79c969cd44
--- /dev/null
+++ b/sound/soc/codecs/nau8822.h
@@ -0,0 +1,204 @@
+/*
+ * nau8822.h  --  NAU8822 Soc Audio Codec driver
+ *
+ * Author: David Lin <ctlin0@nuvoton.com>
+ * Co-author: John Hsu <kchsu0@nuvoton.com>
+ * Co-author: Seven Li <wtli@nuvoton.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __NAU8822_H__
+#define __NAU8822_H__
+
+#define NAU8822_REG_RESET			0x00
+#define NAU8822_REG_POWER_MANAGEMENT_1		0x01
+#define NAU8822_REG_POWER_MANAGEMENT_2		0x02
+#define NAU8822_REG_POWER_MANAGEMENT_3		0x03
+#define NAU8822_REG_AUDIO_INTERFACE		0x04
+#define NAU8822_REG_COMPANDING_CONTROL		0x05
+#define NAU8822_REG_CLOCKING			0x06
+#define NAU8822_REG_ADDITIONAL_CONTROL		0x07
+#define NAU8822_REG_GPIO_CONTROL		0x08
+#define NAU8822_REG_JACK_DETECT_CONTROL_1	0x09
+#define NAU8822_REG_DAC_CONTROL			0x0A
+#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME	0x0B
+#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME	0x0C
+#define NAU8822_REG_JACK_DETECT_CONTROL_2	0x0D
+#define NAU8822_REG_ADC_CONTROL			0x0E
+#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME	0x0F
+#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME	0x10
+#define NAU8822_REG_EQ1				0x12
+#define NAU8822_REG_EQ2				0x13
+#define NAU8822_REG_EQ3				0x14
+#define NAU8822_REG_EQ4				0x15
+#define NAU8822_REG_EQ5				0x16
+#define NAU8822_REG_DAC_LIMITER_1		0x18
+#define NAU8822_REG_DAC_LIMITER_2		0x19
+#define NAU8822_REG_NOTCH_FILTER_1		0x1B
+#define NAU8822_REG_NOTCH_FILTER_2		0x1C
+#define NAU8822_REG_NOTCH_FILTER_3		0x1D
+#define NAU8822_REG_NOTCH_FILTER_4		0x1E
+#define NAU8822_REG_ALC_CONTROL_1		0x20
+#define NAU8822_REG_ALC_CONTROL_2		0x21
+#define NAU8822_REG_ALC_CONTROL_3		0x22
+#define NAU8822_REG_NOISE_GATE			0x23
+#define NAU8822_REG_PLL_N			0x24
+#define NAU8822_REG_PLL_K1			0x25
+#define NAU8822_REG_PLL_K2			0x26
+#define NAU8822_REG_PLL_K3			0x27
+#define NAU8822_REG_3D_CONTROL			0x29
+#define NAU8822_REG_RIGHT_SPEAKER_CONTROL	0x2B
+#define NAU8822_REG_INPUT_CONTROL		0x2C
+#define NAU8822_REG_LEFT_INP_PGA_CONTROL	0x2D
+#define NAU8822_REG_RIGHT_INP_PGA_CONTROL	0x2E
+#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL	0x2F
+#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL	0x30
+#define NAU8822_REG_OUTPUT_CONTROL		0x31
+#define NAU8822_REG_LEFT_MIXER_CONTROL		0x32
+#define NAU8822_REG_RIGHT_MIXER_CONTROL		0x33
+#define NAU8822_REG_LHP_VOLUME			0x34
+#define NAU8822_REG_RHP_VOLUME			0x35
+#define NAU8822_REG_LSPKOUT_VOLUME		0x36
+#define NAU8822_REG_RSPKOUT_VOLUME		0x37
+#define NAU8822_REG_AUX2_MIXER			0x38
+#define NAU8822_REG_AUX1_MIXER			0x39
+#define NAU8822_REG_POWER_MANAGEMENT_4		0x3A
+#define NAU8822_REG_LEFT_TIME_SLOT		0x3B
+#define NAU8822_REG_MISC			0x3C
+#define NAU8822_REG_RIGHT_TIME_SLOT		0x3D
+#define NAU8822_REG_DEVICE_REVISION		0x3E
+#define NAU8822_REG_DEVICE_ID			0x3F
+#define NAU8822_REG_DAC_DITHER			0x41
+#define NAU8822_REG_ALC_ENHANCE_1		0x46
+#define NAU8822_REG_ALC_ENHANCE_2		0x47
+#define NAU8822_REG_192KHZ_SAMPLING		0x48
+#define NAU8822_REG_MISC_CONTROL		0x49
+#define NAU8822_REG_INPUT_TIEOFF		0x4A
+#define NAU8822_REG_POWER_REDUCTION		0x4B
+#define NAU8822_REG_AGC_PEAK2PEAK		0x4C
+#define NAU8822_REG_AGC_PEAK_DETECT		0x4D
+#define NAU8822_REG_AUTOMUTE_CONTROL		0x4E
+#define NAU8822_REG_OUTPUT_TIEOFF		0x4F
+#define NAU8822_REG_MAX_REGISTER		NAU8822_REG_OUTPUT_TIEOFF
+
+/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */
+#define NAU8822_REFIMP_MASK			0x3
+#define NAU8822_REFIMP_80K			0x1
+#define NAU8822_REFIMP_300K			0x2
+#define NAU8822_REFIMP_3K			0x3
+#define NAU8822_IOBUF_EN			(0x1 << 2)
+#define NAU8822_ABIAS_EN			(0x1 << 3)
+
+/* NAU8822_REG_AUDIO_INTERFACE (0x4) */
+#define NAU8822_AIFMT_MASK			(0x3 << 3)
+#define NAU8822_WLEN_MASK			(0x3 << 5)
+#define NAU8822_WLEN_20				(0x1 << 5)
+#define NAU8822_WLEN_24				(0x2 << 5)
+#define NAU8822_WLEN_32				(0x3 << 5)
+#define NAU8822_LRP_MASK			(0x1 << 7)
+#define NAU8822_BCLKP_MASK			(0x1 << 8)
+
+/* NAU8822_REG_COMPANDING_CONTROL (0x5) */
+#define NAU8822_ADDAP_SFT			0
+#define NAU8822_ADCCM_SFT			1
+#define NAU8822_DACCM_SFT			3
+
+/* NAU8822_REG_CLOCKING (0x6) */
+#define NAU8822_CLKIOEN_MASK			0x1
+#define NAU8822_MCLKSEL_SFT			5
+#define NAU8822_MCLKSEL_MASK			(0x7 << 5)
+#define NAU8822_BCLKSEL_SFT			2
+#define NAU8822_BCLKSEL_MASK			(0x7 << 2)
+#define NAU8822_CLKM_MASK			(0x1 << 8)
+#define NAU8822_CLKM_MCLK			(0x0 << 8)
+#define NAU8822_CLKM_PLL			(0x1 << 8)
+
+/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */
+#define NAU8822_SMPLR_SFT			1
+#define NAU8822_SMPLR_MASK			(0x7 << 1)
+#define NAU8822_SMPLR_48K			(0x0 << 1)
+#define NAU8822_SMPLR_32K			(0x1 << 1)
+#define NAU8822_SMPLR_24K			(0x2 << 1)
+#define NAU8822_SMPLR_16K			(0x3 << 1)
+#define NAU8822_SMPLR_12K			(0x4 << 1)
+#define NAU8822_SMPLR_8K			(0x5 << 1)
+
+/* NAU8822_REG_EQ1 (0x12) */
+#define NAU8822_EQ1GC_SFT			0
+#define NAU8822_EQ1CF_SFT			5
+#define NAU8822_EQM_SFT				8
+
+/* NAU8822_REG_EQ2 (0x13) */
+#define NAU8822_EQ2GC_SFT			0
+#define NAU8822_EQ2CF_SFT			5
+#define NAU8822_EQ2BW_SFT			8
+
+/* NAU8822_REG_EQ3 (0x14) */
+#define NAU8822_EQ3GC_SFT			0
+#define NAU8822_EQ3CF_SFT			5
+#define NAU8822_EQ3BW_SFT			8
+
+/* NAU8822_REG_EQ4 (0x15) */
+#define NAU8822_EQ4GC_SFT			0
+#define NAU8822_EQ4CF_SFT			5
+#define NAU8822_EQ4BW_SFT			8
+
+/* NAU8822_REG_EQ5 (0x16) */
+#define NAU8822_EQ5GC_SFT			0
+#define NAU8822_EQ5CF_SFT			5
+
+/* NAU8822_REG_ALC_CONTROL_1 (0x20) */
+#define NAU8822_ALCMINGAIN_SFT			0
+#define NAU8822_ALCMXGAIN_SFT			3
+#define NAU8822_ALCEN_SFT			7
+
+/* NAU8822_REG_ALC_CONTROL_2 (0x21) */
+#define NAU8822_ALCSL_SFT			0
+#define NAU8822_ALCHT_SFT			4
+
+/* NAU8822_REG_ALC_CONTROL_3 (0x22) */
+#define NAU8822_ALCATK_SFT			0
+#define NAU8822_ALCDCY_SFT			4
+#define NAU8822_ALCM_SFT			8
+
+/* NAU8822_REG_PLL_N (0x24) */
+#define NAU8822_PLLMCLK_DIV2			(0x1 << 4)
+#define NAU8822_PLLN_MASK			0xF
+
+#define NAU8822_PLLK1_SFT			18
+#define NAU8822_PLLK1_MASK			0x3F
+
+/* NAU8822_REG_PLL_K2 (0x26) */
+#define NAU8822_PLLK2_SFT			9
+#define NAU8822_PLLK2_MASK			0x1FF
+
+/* NAU8822_REG_PLL_K3 (0x27) */
+#define NAU8822_PLLK3_MASK			0x1FF
+
+/* System Clock Source */
+enum {
+	NAU8822_CLK_MCLK,
+	NAU8822_CLK_PLL,
+};
+
+struct nau8822_pll {
+	int pre_factor;
+	int mclk_scaler;
+	int pll_frac;
+	int pll_int;
+};
+
+/* Codec Private Data */
+struct nau8822 {
+	struct device *dev;
+	struct regmap *regmap;
+	int mclk_idx;
+	struct nau8822_pll pll;
+	int sysclk;
+	int div_id;
+};
+
+#endif	/* __NAU8822_H__ */
diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c
index 690c26e7389e..809b7e9f03ca 100644
--- a/sound/soc/codecs/pcm186x.c
+++ b/sound/soc/codecs/pcm186x.c
@@ -401,7 +401,8 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format)
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		priv->tdm_offset += 1;
-		/* Fall through... DSP_A uses the same basic config as DSP_B
+		/* fall through */
+		/* DSP_A uses the same basic config as DSP_B
 		 * except we need to shift the TDM output by one BCK cycle
 		 */
 	case SND_SOC_DAIFMT_DSP_B:
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
new file mode 100644
index 000000000000..cdc8314882bc
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -0,0 +1,60 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 I2C driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct pcm3060_priv *priv;
+
+	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, priv);
+
+	priv->regmap = devm_regmap_init_i2c(i2c, &pcm3060_regmap);
+	if (IS_ERR(priv->regmap))
+		return PTR_ERR(priv->regmap);
+
+	return pcm3060_probe(&i2c->dev);
+}
+
+static const struct i2c_device_id pcm3060_i2c_id[] = {
+	{ .name = "pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(i2c, pcm3060_i2c_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+	{ .compatible = "ti,pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct i2c_driver pcm3060_i2c_driver = {
+	.driver = {
+		.name = "pcm3060",
+#ifdef CONFIG_OF
+		.of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+	},
+	.id_table = pcm3060_i2c_id,
+	.probe = pcm3060_i2c_probe,
+};
+
+module_i2c_driver(pcm3060_i2c_driver);
+
+MODULE_DESCRIPTION("PCM3060 I2C driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
new file mode 100644
index 000000000000..f6f19fa80932
--- /dev/null
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -0,0 +1,59 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 SPI driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "pcm3060.h"
+
+static int pcm3060_spi_probe(struct spi_device *spi)
+{
+	struct pcm3060_priv *priv;
+
+	priv = devm_kzalloc(&spi->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	spi_set_drvdata(spi, priv);
+
+	priv->regmap = devm_regmap_init_spi(spi, &pcm3060_regmap);
+	if (IS_ERR(priv->regmap))
+		return PTR_ERR(priv->regmap);
+
+	return pcm3060_probe(&spi->dev);
+}
+
+static const struct spi_device_id pcm3060_spi_id[] = {
+	{ .name = "pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, pcm3060_spi_id);
+
+#ifdef CONFIG_OF
+static const struct of_device_id pcm3060_of_match[] = {
+	{ .compatible = "ti,pcm3060" },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, pcm3060_of_match);
+#endif /* CONFIG_OF */
+
+static struct spi_driver pcm3060_spi_driver = {
+	.driver = {
+		.name = "pcm3060",
+#ifdef CONFIG_OF
+		.of_match_table = pcm3060_of_match,
+#endif /* CONFIG_OF */
+	},
+	.id_table = pcm3060_spi_id,
+	.probe = pcm3060_spi_probe,
+};
+
+module_spi_driver(pcm3060_spi_driver);
+
+MODULE_DESCRIPTION("PCM3060 SPI driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
new file mode 100644
index 000000000000..494d9d662be8
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.c
@@ -0,0 +1,295 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// PCM3060 codec driver
+//
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "pcm3060.h"
+
+/* dai */
+
+static int pcm3060_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+			      unsigned int freq, int dir)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+
+	if (dir != SND_SOC_CLOCK_IN) {
+		dev_err(comp->dev, "unsupported sysclock dir: %d\n", dir);
+		return -EINVAL;
+	}
+
+	priv->dai[dai->id].sclk_freq = freq;
+
+	return 0;
+}
+
+static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+	unsigned int reg;
+	unsigned int val;
+
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		dev_err(comp->dev, "unsupported DAI polarity: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		priv->dai[dai->id].is_master = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		priv->dai[dai->id].is_master = false;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val = PCM3060_REG_FMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val = PCM3060_REG_FMT_RJ;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val = PCM3060_REG_FMT_LJ;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported DAI format: 0x%x\n", fmt);
+		return -EINVAL;
+	}
+
+	if (dai->id == PCM3060_DAI_ID_DAC)
+		reg = PCM3060_REG67;
+	else
+		reg = PCM3060_REG72;
+
+	regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_FMT, val);
+
+	return 0;
+}
+
+static int pcm3060_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *comp = dai->component;
+	struct pcm3060_priv *priv = snd_soc_component_get_drvdata(comp);
+	unsigned int rate;
+	unsigned int ratio;
+	unsigned int reg;
+	unsigned int val;
+
+	if (!priv->dai[dai->id].is_master) {
+		val = PCM3060_REG_MS_S;
+		goto val_ready;
+	}
+
+	rate = params_rate(params);
+	if (!rate) {
+		dev_err(comp->dev, "rate is not configured\n");
+		return -EINVAL;
+	}
+
+	ratio = priv->dai[dai->id].sclk_freq / rate;
+
+	switch (ratio) {
+	case 768:
+		val = PCM3060_REG_MS_M768;
+		break;
+	case 512:
+		val = PCM3060_REG_MS_M512;
+		break;
+	case 384:
+		val = PCM3060_REG_MS_M384;
+		break;
+	case 256:
+		val = PCM3060_REG_MS_M256;
+		break;
+	case 192:
+		val = PCM3060_REG_MS_M192;
+		break;
+	case 128:
+		val = PCM3060_REG_MS_M128;
+		break;
+	default:
+		dev_err(comp->dev, "unsupported ratio: %d\n", ratio);
+		return -EINVAL;
+	}
+
+val_ready:
+	if (dai->id == PCM3060_DAI_ID_DAC)
+		reg = PCM3060_REG67;
+	else
+		reg = PCM3060_REG72;
+
+	regmap_update_bits(priv->regmap, reg, PCM3060_REG_MASK_MS, val);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops pcm3060_dai_ops = {
+	.set_sysclk = pcm3060_set_sysclk,
+	.set_fmt = pcm3060_set_fmt,
+	.hw_params = pcm3060_hw_params,
+};
+
+#define PCM3060_DAI_RATES_ADC	(SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | \
+				 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+				 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PCM3060_DAI_RATES_DAC	(PCM3060_DAI_RATES_ADC | \
+				 SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+static struct snd_soc_dai_driver pcm3060_dai[] = {
+	{
+		.name = "pcm3060-dac",
+		.id = PCM3060_DAI_ID_DAC,
+		.playback = {
+			.stream_name = "Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = PCM3060_DAI_RATES_DAC,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops = &pcm3060_dai_ops,
+	},
+	{
+		.name = "pcm3060-adc",
+		.id = PCM3060_DAI_ID_ADC,
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = PCM3060_DAI_RATES_ADC,
+			.formats = SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops = &pcm3060_dai_ops,
+	},
+};
+
+/* dapm */
+
+static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1);
+
+static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
+	SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume",
+			       PCM3060_REG65, PCM3060_REG66, 0,
+			       PCM3060_REG_AT2_MIN, PCM3060_REG_AT2_MAX,
+			       0, pcm3060_dapm_tlv),
+	SOC_DOUBLE("Master Playback Switch", PCM3060_REG68,
+		   PCM3060_REG_SHIFT_MUT21, PCM3060_REG_SHIFT_MUT22, 1, 1),
+
+	SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume",
+			       PCM3060_REG70, PCM3060_REG71, 0,
+			       PCM3060_REG_AT1_MIN, PCM3060_REG_AT1_MAX,
+			       0, pcm3060_dapm_tlv),
+	SOC_DOUBLE("Master Capture Switch", PCM3060_REG73,
+		   PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1),
+};
+
+static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("OUTL+"),
+	SND_SOC_DAPM_OUTPUT("OUTR+"),
+	SND_SOC_DAPM_OUTPUT("OUTL-"),
+	SND_SOC_DAPM_OUTPUT("OUTR-"),
+
+	SND_SOC_DAPM_INPUT("INL"),
+	SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
+	{ "OUTL+", NULL, "Playback" },
+	{ "OUTR+", NULL, "Playback" },
+	{ "OUTL-", NULL, "Playback" },
+	{ "OUTR-", NULL, "Playback" },
+
+	{ "Capture", NULL, "INL" },
+	{ "Capture", NULL, "INR" },
+};
+
+/* soc component */
+
+static const struct snd_soc_component_driver pcm3060_soc_comp_driver = {
+	.controls = pcm3060_dapm_controls,
+	.num_controls = ARRAY_SIZE(pcm3060_dapm_controls),
+	.dapm_widgets = pcm3060_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(pcm3060_dapm_widgets),
+	.dapm_routes = pcm3060_dapm_map,
+	.num_dapm_routes = ARRAY_SIZE(pcm3060_dapm_map),
+};
+
+/* regmap */
+
+static bool pcm3060_reg_writeable(struct device *dev, unsigned int reg)
+{
+	return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_readable(struct device *dev, unsigned int reg)
+{
+	return (reg >= PCM3060_REG64);
+}
+
+static bool pcm3060_reg_volatile(struct device *dev, unsigned int reg)
+{
+	/* PCM3060_REG64 is volatile */
+	return (reg == PCM3060_REG64);
+}
+
+static const struct reg_default pcm3060_reg_defaults[] = {
+	{ PCM3060_REG64,  0xF0 },
+	{ PCM3060_REG65,  0xFF },
+	{ PCM3060_REG66,  0xFF },
+	{ PCM3060_REG67,  0x00 },
+	{ PCM3060_REG68,  0x00 },
+	{ PCM3060_REG69,  0x00 },
+	{ PCM3060_REG70,  0xD7 },
+	{ PCM3060_REG71,  0xD7 },
+	{ PCM3060_REG72,  0x00 },
+	{ PCM3060_REG73,  0x00 },
+};
+
+const struct regmap_config pcm3060_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.writeable_reg = pcm3060_reg_writeable,
+	.readable_reg = pcm3060_reg_readable,
+	.volatile_reg = pcm3060_reg_volatile,
+	.max_register = PCM3060_REG73,
+	.reg_defaults = pcm3060_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(pcm3060_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(pcm3060_regmap);
+
+/* device */
+
+int pcm3060_probe(struct device *dev)
+{
+	int rc;
+
+	rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver,
+					     pcm3060_dai,
+					     ARRAY_SIZE(pcm3060_dai));
+	if (rc) {
+		dev_err(dev, "failed to register component, rc=%d\n", rc);
+		return rc;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL(pcm3060_probe);
+
+MODULE_DESCRIPTION("PCM3060 codec driver");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
new file mode 100644
index 000000000000..fd89a68aa8a7
--- /dev/null
+++ b/sound/soc/codecs/pcm3060.h
@@ -0,0 +1,88 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * PCM3060 codec driver
+ *
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ */
+
+#ifndef _SND_SOC_PCM3060_H
+#define _SND_SOC_PCM3060_H
+
+#include <linux/device.h>
+#include <linux/regmap.h>
+
+extern const struct regmap_config pcm3060_regmap;
+
+#define PCM3060_DAI_ID_DAC	0
+#define PCM3060_DAI_ID_ADC	1
+#define PCM3060_DAI_IDS_NUM	2
+
+struct pcm3060_priv_dai {
+	bool is_master;
+	unsigned int sclk_freq;
+};
+
+struct pcm3060_priv {
+	struct regmap *regmap;
+	struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM];
+};
+
+int pcm3060_probe(struct device *dev);
+int pcm3060_remove(struct device *dev);
+
+/* registers */
+
+#define PCM3060_REG64			0x40
+#define PCM3060_REG_MRST		0x80
+#define PCM3060_REG_SRST		0x40
+#define PCM3060_REG_ADPSV		0x20
+#define PCM3060_REG_DAPSV		0x10
+#define PCM3060_REG_SE			0x01
+
+#define PCM3060_REG65			0x41
+#define PCM3060_REG66			0x42
+#define PCM3060_REG_AT2_MIN		0x36
+#define PCM3060_REG_AT2_MAX		0xFF
+
+#define PCM3060_REG67			0x43
+#define PCM3060_REG72			0x48
+#define PCM3060_REG_CSEL		0x80
+#define PCM3060_REG_MASK_MS		0x70
+#define PCM3060_REG_MS_S		0x00
+#define PCM3060_REG_MS_M768		(0x01 << 4)
+#define PCM3060_REG_MS_M512		(0x02 << 4)
+#define PCM3060_REG_MS_M384		(0x03 << 4)
+#define PCM3060_REG_MS_M256		(0x04 << 4)
+#define PCM3060_REG_MS_M192		(0x05 << 4)
+#define PCM3060_REG_MS_M128		(0x06 << 4)
+#define PCM3060_REG_MASK_FMT		0x03
+#define PCM3060_REG_FMT_I2S		0x00
+#define PCM3060_REG_FMT_LJ		0x01
+#define PCM3060_REG_FMT_RJ		0x02
+
+#define PCM3060_REG68			0x44
+#define PCM3060_REG_OVER		0x40
+#define PCM3060_REG_DREV2		0x04
+#define PCM3060_REG_SHIFT_MUT21	0x00
+#define PCM3060_REG_SHIFT_MUT22	0x01
+
+#define PCM3060_REG69			0x45
+#define PCM3060_REG_FLT		0x80
+#define PCM3060_REG_MASK_DMF		0x60
+#define PCM3060_REG_DMC		0x10
+#define PCM3060_REG_ZREV		0x02
+#define PCM3060_REG_AZRO		0x01
+
+#define PCM3060_REG70			0x46
+#define PCM3060_REG71			0x47
+#define PCM3060_REG_AT1_MIN		0x0E
+#define PCM3060_REG_AT1_MAX		0xFF
+
+#define PCM3060_REG73			0x49
+#define PCM3060_REG_ZCDD		0x10
+#define PCM3060_REG_BYP		0x08
+#define PCM3060_REG_DREV1		0x04
+#define PCM3060_REG_SHIFT_MUT11	0x00
+#define PCM3060_REG_SHIFT_MUT12	0x01
+
+#endif /* _SND_SOC_PCM3060_H */
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 3356c91f55b0..52cc950c9fd1 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -33,6 +33,8 @@
 #define PCM3168A_FMT_RIGHT_J_16		0x3
 #define PCM3168A_FMT_DSP_A		0x4
 #define PCM3168A_FMT_DSP_B		0x5
+#define PCM3168A_FMT_I2S_TDM		0x6
+#define PCM3168A_FMT_LEFT_J_TDM		0x7
 #define PCM3168A_FMT_DSP_MASK		0x4
 
 #define PCM3168A_NUM_SUPPLIES 6
@@ -401,9 +403,11 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	bool tx, master_mode;
 	u32 val, mask, shift, reg;
 	unsigned int rate, fmt, ratio, max_ratio;
+	unsigned int chan;
 	int i, min_frame_size;
 
 	rate = params_rate(params);
+	chan = params_channels(params);
 
 	ratio = pcm3168a->sysclk / rate;
 
@@ -456,6 +460,21 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	/* for TDM */
+	if (chan > 2) {
+		switch (fmt) {
+		case PCM3168A_FMT_I2S:
+			fmt = PCM3168A_FMT_I2S_TDM;
+			break;
+		case PCM3168A_FMT_LEFT_J:
+			fmt = PCM3168A_FMT_LEFT_J_TDM;
+			break;
+		default:
+			dev_err(component->dev, "TDM is supported under I2S/Left_J only\n");
+			return -EINVAL;
+		}
+	}
+
 	if (master_mode)
 		val = ((i + 1) << shift);
 	else
@@ -476,7 +495,69 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int pcm3168a_startup(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int fmt;
+	unsigned int sample_min;
+	unsigned int channel_max;
+
+	if (tx)
+		fmt = pcm3168a->dac_fmt;
+	else
+		fmt = pcm3168a->adc_fmt;
+
+	/*
+	 * Available Data Bits
+	 *
+	 * RIGHT_J : 24 / 16
+	 * LEFT_J  : 24
+	 * I2S     : 24
+	 *
+	 * TDM available
+	 *
+	 * I2S
+	 * LEFT_J
+	 */
+	switch (fmt) {
+	case PCM3168A_FMT_RIGHT_J:
+		sample_min  = 16;
+		channel_max =  2;
+		break;
+	case PCM3168A_FMT_LEFT_J:
+		sample_min  = 24;
+		if (tx)
+			channel_max = 8;
+		else
+			channel_max = 6;
+		break;
+	case PCM3168A_FMT_I2S:
+		sample_min  = 24;
+		if (tx)
+			channel_max = 8;
+		else
+			channel_max = 6;
+		break;
+	default:
+		sample_min  = 24;
+		channel_max =  2;
+	}
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+				     sample_min, 32);
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS,
+				     2, channel_max);
+
+	return 0;
+}
 static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
+	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_dac,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params,
@@ -484,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = {
 };
 
 static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = {
+	.startup	= pcm3168a_startup,
 	.set_fmt	= pcm3168a_set_dai_fmt_adc,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index d88e67341083..0ef966d56bac 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -755,6 +755,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 		break;
 	default:
 		dev_warn(component->dev, "invalid pll source, use BCLK\n");
+		/* fall through */
 	case RT274_PLL2_S_BCLK:
 		snd_soc_component_update_bits(component, RT274_PLL2_CTRL,
 				RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK);
@@ -782,6 +783,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
 			break;
 		default:
 			dev_warn(component->dev, "invalid freq_in, assume 4.8M\n");
+			/* fall through */
 		case 100:
 			snd_soc_component_write(component, 0x7a, 0xaab6);
 			snd_soc_component_write(component, 0x7b, 0x0301);
diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c
index 6478d10c4f4a..4d46f4567c3a 100644
--- a/sound/soc/codecs/rt5514-spi.c
+++ b/sound/soc/codecs/rt5514-spi.c
@@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work)
 
 	runtime = rt5514_dsp->substream->runtime;
 	period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
+	if (!period_bytes) {
+		schedule_delayed_work(&rt5514_dsp->copy_work, 5);
+		goto done;
+	}
+
+	if (rt5514_dsp->buf_size % period_bytes)
+		rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
+			period_bytes;
 
 	if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) {
 		rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf,
@@ -149,13 +157,11 @@ done:
 
 static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
 {
-	size_t period_bytes;
 	u8 buf[8];
 
 	if (!rt5514_dsp->substream)
 		return;
 
-	period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream);
 	rt5514_dsp->get_size = 0;
 
 	/**
@@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp)
 
 	rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base;
 
-	if (rt5514_dsp->buf_size % period_bytes)
-		rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) *
-			period_bytes;
-
 	if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit &&
 		rt5514_dsp->buf_rp && rt5514_dsp->buf_size)
 		schedule_delayed_work(&rt5514_dsp->copy_work, 0);
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 985852fd9723..b613103d801b 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -10,7 +10,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 9bd24ad42240..2444fad7c2df 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -72,6 +72,7 @@ struct rt5663_priv {
 static const struct reg_sequence rt5663_patch_list[] = {
 	{ 0x002a, 0x8020 },
 	{ 0x0086, 0x0028 },
+	{ 0x0100, 0xa020 },
 	{ 0x0117, 0x0f28 },
 	{ 0x02fb, 0x8089 },
 };
@@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x00fd, 0x0001 },
 	{ 0x00fe, 0x10ec },
 	{ 0x00ff, 0x6406 },
-	{ 0x0100, 0xa0a0 },
+	{ 0x0100, 0xa020 },
 	{ 0x0108, 0x4444 },
 	{ 0x0109, 0x4444 },
 	{ 0x010a, 0xaaaa },
@@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				0x8000);
 			snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000,
 				0x3000);
+			snd_soc_component_update_bits(component,
+				RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080);
 		}
 		break;
 
@@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w,
 				RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN);
 			snd_soc_component_update_bits(component,
 				RT5663_DACREF_LDO, 0x3e0e, 0);
+			snd_soc_component_update_bits(component,
+				RT5663_DIG_VOL_ZCD, 0x00c0, 0);
 		}
 		break;
 
diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c
index 3c19d03f2446..85ba04d6e7ae 100644
--- a/sound/soc/codecs/rt5668.c
+++ b/sound/soc/codecs/rt5668.c
@@ -2587,17 +2587,10 @@ static int rt5668_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
+	return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668,
 			rt5668_dai, ARRAY_SIZE(rt5668_dai));
 }
 
-static int rt5668_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-
-	return 0;
-}
-
 static void rt5668_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt5668_priv *rt5668 = i2c_get_clientdata(client);
@@ -2628,7 +2621,6 @@ static struct i2c_driver rt5668_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5668_acpi_match),
 	},
 	.probe = rt5668_i2c_probe,
-	.remove = rt5668_i2c_remove,
 	.shutdown = rt5668_i2c_shutdown,
 	.id_table = rt5668_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 732ef928b25d..455fe7cff700 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2877,6 +2877,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
 	},
 	{
 		.callback = rt5670_quirk_cb,
+		.ident = "Lenovo Thinkpad Tablet 8",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+		},
+		.driver_data = (unsigned long *)(RT5670_DMIC_EN |
+						 RT5670_DMIC2_INR |
+						 RT5670_DEV_GPIO |
+						 RT5670_JD_MODE1),
+	},
+	{
+		.callback = rt5670_quirk_cb,
 		.ident = "Lenovo Thinkpad Tablet 10",
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index bd51f3655ee3..84501c2020c7 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -18,7 +18,6 @@
 #include <linux/interrupt.h>
 #include <linux/irq.h>
 #include <linux/slab.h>
-#include <linux/gpio.h>
 #include <linux/sched.h>
 #include <linux/uaccess.h>
 #include <linux/regulator/consumer.h>
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index afe7d5b19313..340f90497d07 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -67,7 +67,8 @@ struct rt5682_priv {
 };
 
 static const struct reg_sequence patch_list[] = {
-	{0x01c1, 0x1000},
+	{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
+	{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
 };
 
 static const struct reg_default rt5682_reg[] = {
@@ -749,7 +750,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg)
 	}
 }
 
-static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0);
 static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
@@ -1108,10 +1108,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
 }
 
 static const struct snd_kcontrol_new rt5682_snd_controls[] = {
-	/* Headphone Output Volume */
-	SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN,
-		RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv),
-
 	/* DAC Digital Volume */
 	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
 		RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv),
@@ -1437,6 +1433,28 @@ static const struct snd_kcontrol_new hpor_switch =
 	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1,
 					RT5682_R_MUTE_SFT, 1, 1);
 
+static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -1449,10 +1467,10 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_HP_LOGIC_CTRL_2, 0x0012);
 		snd_soc_component_write(component,
 			RT5682_HP_CTRL_2, 0x6000);
-		snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1,
-			RT5682_NG2_EN_MASK, RT5682_NG2_EN);
 		snd_soc_component_update_bits(component,
 			RT5682_DEPOP_1, 0x60, 0x60);
+		snd_soc_component_update_bits(component,
+			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080);
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
@@ -1460,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_DEPOP_1, 0x60, 0x0);
 		snd_soc_component_write(component,
 			RT5682_HP_CTRL_2, 0x0000);
+		snd_soc_component_update_bits(component,
+			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000);
 		break;
 
 	default:
@@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1,
 		RT5682_PWR_HA_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1,
-		RT5682_PUMP_EN_SFT, 0, NULL, 0),
+		RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event,
+		SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1,
 		RT5682_CAPLESS_EN_SFT, 0, NULL, 0),
 
@@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
 	{"HP Amp", NULL, "Charge Pump"},
 	{"HP Amp", NULL, "CLKDET SYS"},
 	{"HP Amp", NULL, "CBJ Power"},
+	{"HP Amp", NULL, "Vref1"},
 	{"HP Amp", NULL, "Vref2"},
 	{"HPOL Playback", "Switch", "HP Amp"},
 	{"HPOR Playback", "Switch", "HP Amp"},
@@ -2451,30 +2473,23 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 	mutex_lock(&rt5682->calibrate_mutex);
 
 	rt5682_reset(rt5682->regmap);
-	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
 	usleep_range(15000, 20000);
-	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf);
-	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
-	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001);
-	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
-	regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080);
-	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040);
-	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300);
+	regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000);
+	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100);
+	regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800);
 	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000);
-	regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000);
-	regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26);
-	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005);
 	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c);
 	regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d);
-	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f);
-	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321);
 	regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1);
 	regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311);
-	regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000);
-	regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320);
+	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00);
 
 	regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00);
 
@@ -2490,8 +2505,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 		pr_err("HP Calibration Failure\n");
 
 	/* restore settings */
-	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
+	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080);
+	regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000);
 	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
+	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
+	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
 
 	mutex_unlock(&rt5682->calibrate_mutex);
 
@@ -2565,7 +2584,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	rt5682_calibrate(rt5682);
 
-	ret = regmap_register_patch(rt5682->regmap, patch_list,
+	ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
 				    ARRAY_SIZE(patch_list));
 	if (ret != 0)
 		dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret);
@@ -2619,6 +2638,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 			RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK,
 			RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1);
 	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+	regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+			RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+	regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+			RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
 
 	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
 				rt5682_jack_detect_handler);
@@ -2636,11 +2659,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return devm_snd_soc_register_component(&i2c->dev,
-			&soc_component_dev_rt5682,
+	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682,
 			rt5682_dai, ARRAY_SIZE(rt5682_dai));
 }
 
+static int rt5682_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_component(&i2c->dev);
+
+	return 0;
+}
+
 static void rt5682_i2c_shutdown(struct i2c_client *client)
 {
 	struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
@@ -2671,6 +2700,7 @@ static struct i2c_driver rt5682_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
 	},
 	.probe = rt5682_i2c_probe,
+	.remove = rt5682_i2c_remove,
 	.shutdown = rt5682_i2c_shutdown,
 	.id_table = rt5682_i2c_id,
 };
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index 8068140ebe3f..d82a8301fd74 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1214,6 +1214,20 @@
 #define RT5682_JDH_NO_PLUG			(0x1 << 4)
 #define RT5682_JDH_PLUG				(0x0 << 4)
 
+/* Bias current control 8 (0x0111) */
+#define RT5682_HPA_CP_BIAS_CTRL_MASK			(0x3 << 2)
+#define RT5682_HPA_CP_BIAS_2UA			(0x0 << 2)
+#define RT5682_HPA_CP_BIAS_3UA			(0x1 << 2)
+#define RT5682_HPA_CP_BIAS_4UA			(0x2 << 2)
+#define RT5682_HPA_CP_BIAS_6UA			(0x3 << 2)
+
+/* Charge Pump Internal Register1 (0x0125) */
+#define RT5682_CP_CLK_HP_MASK			(0x3 << 4)
+#define RT5682_CP_CLK_HP_100KHZ			(0x0 << 4)
+#define RT5682_CP_CLK_HP_200KHZ			(0x1 << 4)
+#define RT5682_CP_CLK_HP_300KHZ			(0x2 << 4)
+#define RT5682_CP_CLK_HP_600KHZ			(0x3 << 4)
+
 /* Chopper and Clock control for DAC (0x013a)*/
 #define RT5682_CKXEN_DAC1_MASK			(0x1 << 13)
 #define RT5682_CKXEN_DAC1_SFT			13
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 60764f6201b1..add18d6d77da 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component)
 	 * Searching for a suitable index solving this formula:
 	 * idx = 40 * log10(vag_val / lo_cagcntrl) + 15
 	 */
-	vol_quot = (vag * 100) / lo_vag;
+	vol_quot = lo_vag ? (vag * 100) / lo_vag : 0;
 	lo_vol = 0;
 	for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) {
 		if (vol_quot >= vol_quot_table[i])
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index d5035f2f2b2b..f753d2db0a5a 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -21,6 +21,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
@@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = {
 /* codec private data */
 struct sta32x_priv {
 	struct regmap *regmap;
+	struct clk *xti_clk;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
 	struct snd_soc_component *component;
 	struct sta32x_platform_data *pdata;
@@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component)
 	struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component);
 	struct sta32x_platform_data *pdata = sta32x->pdata;
 	int i, ret = 0, thermal = 0;
+
+	sta32x->component = component;
+
+	if (sta32x->xti_clk) {
+		ret = clk_prepare_enable(sta32x->xti_clk);
+		if (ret != 0) {
+			dev_err(component->dev,
+				"Failed to enable clock: %d\n", ret);
+			return ret;
+		}
+	}
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
 				    sta32x->supplies);
 	if (ret != 0) {
@@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component)
 
 	sta32x_watchdog_stop(sta32x);
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
+
+	if (sta32x->xti_clk)
+		clk_disable_unprepare(sta32x->xti_clk);
 }
 
 static const struct snd_soc_component_driver sta32x_component = {
@@ -1038,6 +1055,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x)
 	of_property_read_u8(np, "st,ch3-output-mapping",
 			    &pdata->ch3_output_mapping);
 
+	if (of_get_property(np, "st,fault-detect-recovery", NULL))
+		pdata->fault_detect_recovery = 1;
 	if (of_get_property(np, "st,thermal-warning-recovery", NULL))
 		pdata->thermal_warning_recovery = 1;
 	if (of_get_property(np, "st,thermal-warning-adjustment", NULL))
@@ -1095,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c,
 	}
 #endif
 
+	/* Clock */
+	sta32x->xti_clk = devm_clk_get(dev, "xti");
+	if (IS_ERR(sta32x->xti_clk)) {
+		ret = PTR_ERR(sta32x->xti_clk);
+
+		if (ret == -EPROBE_DEFER)
+			return ret;
+
+		sta32x->xti_clk = NULL;
+	}
+
 	/* GPIOs */
 	sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
 						       GPIOD_OUT_LOW);
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index ae3d032ac35a..6bd0e5d5347f 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 				    int slots, int slot_width)
 {
 	struct snd_soc_component *component = dai->component;
+	struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component);
 	unsigned int first_slot;
 	int ret;
 
@@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	if (ret < 0)
 		goto error_snd_soc_component_update_bits;
 
+	/* Configure TDM slot width. This is only applicable to TAS5722. */
+	switch (tas5720->devtype) {
+	case TAS5722:
+		ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+						    TAS5722_TDM_SLOT_16B,
+						    slot_width == 16 ?
+						    TAS5722_TDM_SLOT_16B : 0);
+		if (ret < 0)
+			goto error_snd_soc_component_update_bits;
+		break;
+	default:
+		break;
+	}
+
 	return 0;
 
 error_snd_soc_component_update_bits:
@@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv,
 );
 
 /*
- * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
- * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
- * as per device datasheet.
+ * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps
+ * depending on the device. Note that setting the gain below -100 dB
+ * (register value <0x7) is effectively a MUTE as per device datasheet.
+ *
+ * Note that for the TAS5722 the digital volume controls are actually split
+ * over two registers, so we need custom getters/setters for access.
  */
-static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0);
+
+static int tas5722_volume_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	unsigned int val;
+
+	snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val);
+	ucontrol->value.integer.value[0] = val << 1;
+
+	snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val);
+	ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB;
+
+	return 0;
+}
+
+static int tas5722_volume_set(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	unsigned int sel = ucontrol->value.integer.value[0];
+
+	snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1);
+	snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG,
+				      TAS5722_VOL_CONTROL_LSB, sel);
+
+	return 0;
+}
 
 static const struct snd_kcontrol_new tas5720_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
-		       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv),
+		       TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv),
+	SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
+		       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
+};
+
+static const struct snd_kcontrol_new tas5722_snd_controls[] = {
+	SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume",
+			   0, 0, 511, 0,
+			   tas5722_volume_get, tas5722_volume_set,
+			   tas5722_dac_tlv),
 	SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG,
 		       TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv),
 };
@@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static const struct snd_soc_component_driver soc_component_dev_tas5722 = {
+	.probe = tas5720_codec_probe,
+	.remove = tas5720_codec_remove,
+	.suspend = tas5720_suspend,
+	.resume = tas5720_resume,
+	.controls = tas5722_snd_controls,
+	.num_controls = ARRAY_SIZE(tas5722_snd_controls),
+	.dapm_widgets = tas5720_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets),
+	.dapm_routes = tas5720_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(tas5720_audio_map),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
 /* PCM rates supported by the TAS5720 driver */
 #define TAS5720_RATES	(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
 			 SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
@@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client,
 
 	dev_set_drvdata(dev, data);
 
-	ret = devm_snd_soc_register_component(&client->dev,
-				     &soc_component_dev_tas5720,
-				     tas5720_dai, ARRAY_SIZE(tas5720_dai));
+	switch (id->driver_data) {
+	case TAS5720:
+		ret = devm_snd_soc_register_component(&client->dev,
+					&soc_component_dev_tas5720,
+					tas5720_dai,
+					ARRAY_SIZE(tas5720_dai));
+		break;
+	case TAS5722:
+		ret = devm_snd_soc_register_component(&client->dev,
+					&soc_component_dev_tas5722,
+					tas5720_dai,
+					ARRAY_SIZE(tas5720_dai));
+		break;
+	default:
+		dev_err(dev, "unexpected private driver data\n");
+		return -EINVAL;
+	}
 	if (ret < 0) {
 		dev_err(dev, "failed to register component: %d\n", ret);
 		return ret;
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index 0d6145549a98..36aebdb8f55c 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -41,6 +41,7 @@ struct tas6424_data {
 	struct regmap *regmap;
 	struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES];
 	struct delayed_work fault_check_work;
+	unsigned int last_cfault;
 	unsigned int last_fault1;
 	unsigned int last_fault2;
 	unsigned int last_warn;
@@ -406,9 +407,54 @@ static void tas6424_fault_check_work(struct work_struct *work)
 	unsigned int reg;
 	int ret;
 
+	ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, &reg);
+	if (ret < 0) {
+		dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret);
+		goto out;
+	}
+
+	if (!reg) {
+		tas6424->last_cfault = reg;
+		goto check_global_fault1_reg;
+	}
+
+	/*
+	 * Only flag errors once for a given occurrence. This is needed as
+	 * the TAS6424 will take time clearing the fault condition internally
+	 * during which we don't want to bombard the system with the same
+	 * error message over and over.
+	 */
+	if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1))
+		dev_crit(dev, "experienced a channel 1 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2))
+		dev_crit(dev, "experienced a channel 2 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3))
+		dev_crit(dev, "experienced a channel 3 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4))
+		dev_crit(dev, "experienced a channel 4 overcurrent fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1))
+		dev_crit(dev, "experienced a channel 1 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2))
+		dev_crit(dev, "experienced a channel 2 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3))
+		dev_crit(dev, "experienced a channel 3 DC fault\n");
+
+	if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4))
+		dev_crit(dev, "experienced a channel 4 DC fault\n");
+
+	/* Store current fault1 value so we can detect any changes next time */
+	tas6424->last_cfault = reg;
+
+check_global_fault1_reg:
 	ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg);
 	if (ret < 0) {
-		dev_err(dev, "failed to read FAULT1 register: %d\n", ret);
+		dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret);
 		goto out;
 	}
 
@@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work)
 		goto check_global_fault2_reg;
 	}
 
-	/*
-	 * Only flag errors once for a given occurrence. This is needed as
-	 * the TAS6424 will take time clearing the fault condition internally
-	 * during which we don't want to bombard the system with the same
-	 * error message over and over.
-	 */
 	if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV))
 		dev_crit(dev, "experienced a PVDD overvoltage fault\n");
 
@@ -453,7 +493,7 @@ static void tas6424_fault_check_work(struct work_struct *work)
 check_global_fault2_reg:
 	ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, &reg);
 	if (ret < 0) {
-		dev_err(dev, "failed to read FAULT2 register: %d\n", ret);
+		dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret);
 		goto out;
 	}
 
@@ -530,7 +570,7 @@ check_warn_reg:
 	/* Store current warn value so we can detect any changes next time */
 	tas6424->last_warn = reg;
 
-	/* Clear any faults by toggling the CLEAR_FAULT control bit */
+	/* Clear any warnings by toggling the CLEAR_FAULT control bit */
 	ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3,
 				TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT);
 	if (ret < 0)
diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h
index b5958c45ed0e..c67a7835ca66 100644
--- a/sound/soc/codecs/tas6424.h
+++ b/sound/soc/codecs/tas6424.h
@@ -116,6 +116,16 @@
 #define TAS6424_LDGBYPASS_MASK		BIT(TAS6424_LDGBYPASS_SHIFT)
 
 /* TAS6424_GLOB_FAULT1_REG */
+#define TAS6424_FAULT_OC_CH1		BIT(7)
+#define TAS6424_FAULT_OC_CH2		BIT(6)
+#define TAS6424_FAULT_OC_CH3		BIT(5)
+#define TAS6424_FAULT_OC_CH4		BIT(4)
+#define TAS6424_FAULT_DC_CH1		BIT(3)
+#define TAS6424_FAULT_DC_CH2		BIT(2)
+#define TAS6424_FAULT_DC_CH3		BIT(1)
+#define TAS6424_FAULT_DC_CH4		BIT(0)
+
+/* TAS6424_GLOB_FAULT1_REG */
 #define TAS6424_FAULT_CLOCK		BIT(4)
 #define TAS6424_FAULT_PVDD_OV		BIT(3)
 #define TAS6424_FAULT_VBAT_OV		BIT(2)
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index bf92d36b8f8a..608ad49ad978 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -167,6 +167,7 @@ struct aic31xx_priv {
 	u8 p_div;
 	int rate_div_line;
 	bool master_dapm_route_applied;
+	int irq;
 };
 
 struct aic31xx_rate_divs {
@@ -1391,6 +1392,69 @@ static const struct acpi_device_id aic31xx_acpi_match[] = {
 MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match);
 #endif
 
+static irqreturn_t aic31xx_irq(int irq, void *data)
+{
+	struct aic31xx_priv *aic31xx = data;
+	struct device *dev = aic31xx->dev;
+	unsigned int value;
+	bool handled = false;
+	int ret;
+
+	ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value);
+	if (ret) {
+		dev_err(dev, "Failed to read interrupt mask: %d\n", ret);
+		goto exit;
+	}
+
+	if (value)
+		handled = true;
+	else
+		goto read_overflow;
+
+	if (value & AIC31XX_HPLSCDETECT)
+		dev_err(dev, "Short circuit on Left output is detected\n");
+	if (value & AIC31XX_HPRSCDETECT)
+		dev_err(dev, "Short circuit on Right output is detected\n");
+	if (value & ~(AIC31XX_HPLSCDETECT |
+		      AIC31XX_HPRSCDETECT))
+		dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value);
+
+read_overflow:
+	ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value);
+	if (ret) {
+		dev_err(dev, "Failed to read overflow flag: %d\n", ret);
+		goto exit;
+	}
+
+	if (value)
+		handled = true;
+	else
+		goto exit;
+
+	if (value & AIC31XX_DAC_OF_LEFT)
+		dev_warn(dev, "Left-channel DAC overflow has occurred\n");
+	if (value & AIC31XX_DAC_OF_RIGHT)
+		dev_warn(dev, "Right-channel DAC overflow has occurred\n");
+	if (value & AIC31XX_DAC_OF_SHIFTER)
+		dev_warn(dev, "DAC barrel shifter overflow has occurred\n");
+	if (value & AIC31XX_ADC_OF)
+		dev_warn(dev, "ADC overflow has occurred\n");
+	if (value & AIC31XX_ADC_OF_SHIFTER)
+		dev_warn(dev, "ADC barrel shifter overflow has occurred\n");
+	if (value & ~(AIC31XX_DAC_OF_LEFT |
+		      AIC31XX_DAC_OF_RIGHT |
+		      AIC31XX_DAC_OF_SHIFTER |
+		      AIC31XX_ADC_OF |
+		      AIC31XX_ADC_OF_SHIFTER))
+		dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value);
+
+exit:
+	if (handled)
+		return IRQ_HANDLED;
+	else
+		return IRQ_NONE;
+}
+
 static int aic31xx_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
@@ -1413,6 +1477,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 	aic31xx->dev = &i2c->dev;
+	aic31xx->irq = i2c->irq;
 
 	aic31xx->codec_type = id->driver_data;
 
@@ -1456,6 +1521,26 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
+	if (aic31xx->irq > 0) {
+		regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1,
+				   AIC31XX_GPIO1_FUNC_MASK,
+				   AIC31XX_GPIO1_INT1 <<
+				   AIC31XX_GPIO1_FUNC_SHIFT);
+
+		regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL,
+			     AIC31XX_SC |
+			     AIC31XX_ENGINE);
+
+		ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq,
+						NULL, aic31xx_irq,
+						IRQF_ONESHOT, "aic31xx-irq",
+						aic31xx);
+		if (ret) {
+			dev_err(aic31xx->dev, "Unable to request IRQ\n");
+			return ret;
+		}
+	}
+
 	if (aic31xx->codec_type & DAC31XX_BIT)
 		return devm_snd_soc_register_component(&i2c->dev,
 				&soc_codec_driver_aic31xx,
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index 0b587585b38b..2636f2c6bc79 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -173,6 +173,13 @@ struct aic31xx_pdata {
 #define AIC31XX_HPRDRVPWRSTATUS_MASK	BIT(1)
 #define AIC31XX_SPRDRVPWRSTATUS_MASK	BIT(0)
 
+/* AIC31XX_OFFLAG */
+#define AIC31XX_DAC_OF_LEFT		BIT(7)
+#define AIC31XX_DAC_OF_RIGHT		BIT(6)
+#define AIC31XX_DAC_OF_SHIFTER		BIT(5)
+#define AIC31XX_ADC_OF			BIT(3)
+#define AIC31XX_ADC_OF_SHIFTER		BIT(1)
+
 /* AIC31XX_INTRDACFLAG */
 #define AIC31XX_HPLSCDETECT		BIT(7)
 #define AIC31XX_HPRSCDETECT		BIT(6)
@@ -191,6 +198,22 @@ struct aic31xx_pdata {
 #define AIC31XX_SC			BIT(3)
 #define AIC31XX_ENGINE			BIT(2)
 
+/* AIC31XX_GPIO1 */
+#define AIC31XX_GPIO1_FUNC_MASK		GENMASK(5, 2)
+#define AIC31XX_GPIO1_FUNC_SHIFT	2
+#define AIC31XX_GPIO1_DISABLED		0x00
+#define AIC31XX_GPIO1_INPUT		0x01
+#define AIC31XX_GPIO1_GPI		0x02
+#define AIC31XX_GPIO1_GPO		0x03
+#define AIC31XX_GPIO1_CLKOUT		0x04
+#define AIC31XX_GPIO1_INT1		0x05
+#define AIC31XX_GPIO1_INT2		0x06
+#define AIC31XX_GPIO1_ADC_WCLK		0x07
+#define AIC31XX_GPIO1_SBCLK		0x08
+#define AIC31XX_GPIO1_SWCLK		0x09
+#define AIC31XX_GPIO1_ADC_MOD_CLK	0x10
+#define AIC31XX_GPIO1_SDOUT		0x11
+
 /* AIC31XX_DACSETUP */
 #define AIC31XX_SOFTSTEP_MASK		GENMASK(1, 0)
 
diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c
index ff85a0bf6170..93d84e5ae2d5 100644
--- a/sound/soc/codecs/tscs454.c
+++ b/sound/soc/codecs/tscs454.c
@@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c,
 	/* Sync pg sel reg with cache */
 	regmap_write(tscs454->regmap, R_PAGESEL, 0x00);
 
-	ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
+	ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454,
 			tscs454_dais, ARRAY_SIZE(tscs454_dais));
 	if (ret) {
 		dev_err(&i2c->dev, "Failed to register component (%d)\n", ret);
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c5ae07234a00..bba330e30162 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg,
 	return regmap_write(wm2000->regmap, reg, value);
 }
 
-static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r)
-{
-	struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
-	unsigned int val;
-	int ret;
-
-	ret = regmap_read(wm2000->regmap, r, &val);
-	if (ret < 0)
-		return -1;
-
-	return val;
-}
-
 static void wm2000_reset(struct wm2000_priv *wm2000)
 {
 	struct i2c_client *i2c = wm2000->i2c;
@@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000)
 static int wm2000_poll_bit(struct i2c_client *i2c,
 			   unsigned int reg, u8 mask)
 {
+	struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c);
 	int timeout = 4000;
-	int val;
+	unsigned int val;
 
-	val = wm2000_read(i2c, reg);
+	regmap_read(wm2000->regmap, reg, &val);
 
 	while (!(val & mask) && --timeout) {
 		msleep(1);
-		val = wm2000_read(i2c, reg);
+		regmap_read(wm2000->regmap, reg, &val);
 	}
 
 	if (timeout == 0)
@@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 {
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 	unsigned long rate;
+	unsigned int val;
 	int ret;
 
 	if (WARN_ON(wm2000->anc_mode != ANC_OFF))
@@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 			     WM2000_MODE_THERMAL_ENABLE);
 	}
 
-	ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret);
+		regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies);
+		return ret;
+	}
 	if (wm2000->speech_clarity)
-		ret |= WM2000_SPEECH_CLARITY;
+		val |= WM2000_SPEECH_CLARITY;
 	else
-		ret &= ~WM2000_SPEECH_CLARITY;
-	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
+		val &= ~WM2000_SPEECH_CLARITY;
+	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val);
 
 	wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
 	wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02);
@@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 	const char *filename;
 	const struct firmware *fw = NULL;
 	int ret, i;
-	int reg;
+	unsigned int reg;
 	u16 id;
 
 	wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL);
@@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 	}
 
 	/* Verify that this is a WM2000 */
-	reg = wm2000_read(i2c, WM2000_REG_ID1);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret);
+		return ret;
+	}
 	id = reg << 8;
-	reg = wm2000_read(i2c, WM2000_REG_ID2);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret);
+		return ret;
+	}
 	id |= reg & 0xff;
 
 	if (id != 0x2000) {
@@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c,
 		goto err_supplies;
 	}
 
-	reg = wm2000_read(i2c, WM2000_REG_REVISON);
+	ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret);
+		return ret;
+	}
 	dev_info(&i2c->dev, "revision %c\n", reg + 'A');
 
 	wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK");
diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c
index 317db9a149a7..cf2cdbece122 100644
--- a/sound/soc/codecs/wm8782.c
+++ b/sound/soc/codecs/wm8782.c
@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
+#include <linux/regulator/consumer.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/ac97_codec.h>
@@ -50,7 +51,51 @@ static struct snd_soc_dai_driver wm8782_dai = {
 	},
 };
 
+/* regulator power supply names */
+static const char *supply_names[] = {
+	"Vdda", /* analog supply, 2.7V - 3.6V */
+	"Vdd",  /* digital supply, 2.7V - 5.5V */
+};
+
+struct wm8782_priv {
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static int wm8782_soc_probe(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+static void wm8782_soc_remove(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+
+#ifdef CONFIG_PM
+static int wm8782_soc_suspend(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
+	return 0;
+}
+
+static int wm8782_soc_resume(struct snd_soc_component *component)
+{
+	struct wm8782_priv *priv = snd_soc_component_get_drvdata(component);
+	return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
+}
+#else
+#define wm8782_soc_suspend      NULL
+#define wm8782_soc_resume       NULL
+#endif /* CONFIG_PM */
+
 static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
+	.probe			= wm8782_soc_probe,
+	.remove			= wm8782_soc_remove,
+	.suspend		= wm8782_soc_suspend,
+	.resume			= wm8782_soc_resume,
 	.dapm_widgets		= wm8782_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(wm8782_dapm_widgets),
 	.dapm_routes		= wm8782_dapm_routes,
@@ -63,6 +108,24 @@ static const struct snd_soc_component_driver soc_component_dev_wm8782 = {
 
 static int wm8782_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
+	struct wm8782_priv *priv;
+	int ret, i;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, priv);
+
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		priv->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
+				      priv->supplies);
+	if (ret < 0)
+		return ret;
+
 	return devm_snd_soc_register_component(&pdev->dev,
 			&soc_component_dev_wm8782, &wm8782_dai, 1);
 }
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 1965635ec07c..2a3e5fbd04e4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -13,7 +13,6 @@
 
 #include <linux/clk.h>
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 43edaf8cd276..593a11960888 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -11,7 +11,6 @@
  */
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ade34c26ad2f..e873baa9e778 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component)
 {
 	struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
 	struct regmap *regmap;
-	int ret;
 
 	if (wm9712->mfd_pdata) {
 		wm9712->ac97 = wm9712->mfd_pdata->ac97;
 		regmap = wm9712->mfd_pdata->regmap;
 	} else {
 #ifdef CONFIG_SND_SOC_AC97_BUS
+		int ret;
+
 		wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID,
 						      WM9712_VENDOR_ID_MASK);
 		if (IS_ERR(wm9712->ac97)) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f61656070225..a53dc174bbf0 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct {
 };
 
 struct wm_adsp_buffer {
-	__be32 X_buf_base;		/* XM base addr of first X area */
-	__be32 X_buf_size;		/* Size of 1st X area in words */
-	__be32 X_buf_base2;		/* XM base addr of 2nd X area */
-	__be32 X_buf_brk;		/* Total X size in words */
-	__be32 Y_buf_base;		/* YM base addr of Y area */
-	__be32 wrap;			/* Total size X and Y in words */
+	__be32 buf1_base;		/* Base addr of first buffer area */
+	__be32 buf1_size;		/* Size of buf1 area in DSP words */
+	__be32 buf2_base;		/* Base addr of 2nd buffer area */
+	__be32 buf1_buf2_size;		/* Size of buf1+buf2 in DSP words */
+	__be32 buf3_base;		/* Base addr of buf3 area */
+	__be32 buf_total_size;		/* Size of buf1+buf2+buf3 in DSP words */
 	__be32 high_water_mark;		/* Point at which IRQ is asserted */
 	__be32 irq_count;		/* bits 1-31 count IRQ assertions */
 	__be32 irq_ack;			/* acked IRQ count, bit 0 enables IRQ */
@@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def {
 static const struct wm_adsp_buffer_region_def default_regions[] = {
 	{
 		.mem_type = WMFW_ADSP2_XM,
-		.base_offset = HOST_BUFFER_FIELD(X_buf_base),
-		.size_offset = HOST_BUFFER_FIELD(X_buf_size),
+		.base_offset = HOST_BUFFER_FIELD(buf1_base),
+		.size_offset = HOST_BUFFER_FIELD(buf1_size),
 	},
 	{
 		.mem_type = WMFW_ADSP2_XM,
-		.base_offset = HOST_BUFFER_FIELD(X_buf_base2),
-		.size_offset = HOST_BUFFER_FIELD(X_buf_brk),
+		.base_offset = HOST_BUFFER_FIELD(buf2_base),
+		.size_offset = HOST_BUFFER_FIELD(buf1_buf2_size),
 	},
 	{
 		.mem_type = WMFW_ADSP2_YM,
-		.base_offset = HOST_BUFFER_FIELD(Y_buf_base),
-		.size_offset = HOST_BUFFER_FIELD(wrap),
+		.base_offset = HOST_BUFFER_FIELD(buf3_base),
+		.size_offset = HOST_BUFFER_FIELD(buf_total_size),
 	},
 };
 
@@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf)
 		region->cumulative_size = offset;
 
 		adsp_dbg(buf->dsp,
-			 "region=%d type=%d base=%04x off=%04x size=%04x\n",
+			 "region=%d type=%d base=%08x off=%08x size=%08x\n",
 			 i, region->mem_type, region->base_addr,
 			 region->offset, region->cumulative_size);
 	}
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index f70db8412c7c..267aee776b2d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -1041,6 +1041,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
 	return error_ppm;
 }
 
+static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp)
+{
+	if (!mcasp->txnumevt)
+		return 0;
+
+	return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET);
+}
+
+static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp)
+{
+	if (!mcasp->rxnumevt)
+		return 0;
+
+	return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET);
+}
+
+static snd_pcm_sframes_t davinci_mcasp_delay(
+			struct snd_pcm_substream *substream,
+			struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 fifo_use;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		fifo_use = davinci_mcasp_tx_delay(mcasp);
+	else
+		fifo_use = davinci_mcasp_rx_delay(mcasp);
+
+	/*
+	 * Divide the used locations with the channel count to get the
+	 * FIFO usage in samples (don't care about partial samples in the
+	 * buffer).
+	 */
+	return fifo_use / substream->runtime->channels;
+}
+
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 					struct snd_pcm_hw_params *params,
 					struct snd_soc_dai *cpu_dai)
@@ -1365,6 +1401,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
 	.startup	= davinci_mcasp_startup,
 	.shutdown	= davinci_mcasp_shutdown,
 	.trigger	= davinci_mcasp_trigger,
+	.delay		= davinci_mcasp_delay,
 	.hw_params	= davinci_mcasp_hw_params,
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
 	.set_clkdiv	= davinci_mcasp_set_clkdiv,
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 1033ac6631b0..01052a0808b0 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -151,7 +151,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream,
 	int ret;
 
 	/* Fetch the Back-End dma_data from DPCM */
-	list_for_each_entry(dpcm, &rtd->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(rtd, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *substream_be;
 		struct snd_soc_dai *dai = be->cpu_dai;
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c1d1d06783e5..57b484768a58 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	esai_priv->pdev = pdev;
-	strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1);
+	snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np);
 
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c
index 7f0fa4b52223..9981668ab590 100644
--- a/sound/soc/fsl/fsl_utils.c
+++ b/sound/soc/fsl/fsl_utils.c
@@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np,
 		of_node_put(dma_channel_np);
 		return ret;
 	}
-	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
-		 (unsigned long long) res.start, dma_channel_np->name);
+	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn",
+		 (unsigned long long) res.start, dma_channel_np);
 
 	iprop = of_get_property(dma_channel_np, "cell-index", NULL);
 	if (!iprop) {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index ec731223cab3..e339f36cea95 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -57,6 +57,7 @@ static int pcm030_fabric_probe(struct platform_device *op)
 	struct device_node *platform_np;
 	struct snd_soc_card *card = &pcm030_card;
 	struct pcm030_audio_data *pdata;
+	struct snd_soc_dai_link *dai_link;
 	int ret;
 	int i;
 
@@ -78,8 +79,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
 		return -ENODEV;
 	}
 
-	for (i = 0; i < card->num_links; i++)
-		card->dai_link[i].platform_of_node = platform_np;
+	for_each_card_prelinks(card, i, dai_link)
+		dai_link->platform_of_node = platform_np;
 
 	ret = request_module("snd-soc-wm9712");
 	if (ret)
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 2094d2c8919f..25c819e402e1 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -25,6 +25,8 @@ struct graph_card_data {
 	struct graph_dai_props {
 		struct asoc_simple_dai cpu_dai;
 		struct asoc_simple_dai codec_dai;
+		struct snd_soc_dai_link_component codecs; /* single codec */
+		struct snd_soc_dai_link_component platform;
 		unsigned int mclk_fs;
 	} *dai_props;
 	unsigned int mclk_fs;
@@ -180,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs);
+	of_property_read_u32(cpu_ep,   "mclk-fs", &dai_props->mclk_fs);
+	of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs);
 
 	ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link);
 	if (ret < 0)
@@ -213,7 +216,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port,
 	ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 						"%s-%s",
 						dai_link->cpu_dai_name,
-						dai_link->codec_dai_name);
+						dai_link->codecs->dai_name);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -299,7 +302,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	struct graph_dai_props *dai_props;
 	struct device *dev = &pdev->dev;
 	struct snd_soc_card *card;
-	int num, ret;
+	int num, ret, i;
 
 	/* Allocate the private data and the DAI link array */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -315,6 +318,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW);
 	if (IS_ERR(priv->pa_gpio)) {
 		ret = PTR_ERR(priv->pa_gpio);
diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c
index 92882e392d6c..b83bb31021a9 100644
--- a/sound/soc/generic/audio-graph-scu-card.c
+++ b/sound/soc/generic/audio-graph-scu-card.c
@@ -25,7 +25,11 @@
 struct graph_card_data {
 	struct snd_soc_card snd_card;
 	struct snd_soc_codec_conf codec_conf;
-	struct asoc_simple_dai *dai_props;
+	struct graph_dai_props {
+		struct asoc_simple_dai dai;
+		struct snd_soc_dai_link_component codecs;
+		struct snd_soc_dai_link_component platform;
+	} *dai_props;
 	struct snd_soc_dai_link *dai_link;
 	struct asoc_simple_card_data adata;
 };
@@ -39,18 +43,18 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+	struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	return asoc_simple_card_clk_enable(dai_props);
+	return asoc_simple_card_clk_enable(&dai_props->dai);
 }
 
 static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num);
+	struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num);
 
-	asoc_simple_card_clk_disable(dai_props);
+	asoc_simple_card_clk_disable(&dai_props->dai);
 }
 
 static const struct snd_soc_ops asoc_graph_card_ops = {
@@ -63,7 +67,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	struct graph_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct graph_dai_props *dai_props;
 	int num = rtd->num;
 
 	dai_link	= graph_priv_to_link(priv, num);
@@ -72,7 +76,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 				rtd->cpu_dai :
 				rtd->codec_dai;
 
-	return asoc_simple_card_init_dai(dai, dai_props);
+	return asoc_simple_card_init_dai(dai, &dai_props->dai);
 }
 
 static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -92,15 +96,18 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 {
 	struct device *dev = graph_priv_to_dev(priv);
 	struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx);
-	struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx);
+	struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx);
 	struct snd_soc_card *card = graph_priv_to_card(priv);
 	int ret;
 
 	if (is_fe) {
+		struct snd_soc_dai_link_component *codecs;
+
 		/* BE is dummy */
-		dai_link->codec_of_node		= NULL;
-		dai_link->codec_dai_name	= "snd-soc-dummy-dai";
-		dai_link->codec_name		= "snd-soc-dummy";
+		codecs			= dai_link->codecs;
+		codecs->of_node		= NULL;
+		codecs->dai_name	= "snd-soc-dummy-dai";
+		codecs->name		= "snd-soc-dummy";
 
 		/* FE settings */
 		dai_link->dynamic		= 1;
@@ -110,7 +117,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 		if (ret)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
@@ -137,23 +144,23 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep,
 		if (ret < 0)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
 		ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 							"be.%s",
-							dai_link->codec_dai_name);
+							dai_link->codecs->dai_name);
 		if (ret < 0)
 			return ret;
 
 		snd_soc_of_parse_audio_prefix(card,
 					      &priv->codec_conf,
-					      dai_link->codec_of_node,
+					      dai_link->codecs->of_node,
 					      "prefix");
 	}
 
-	ret = asoc_simple_card_of_parse_tdm(ep, dai_props);
+	ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai);
 	if (ret)
 		return ret;
 
@@ -331,10 +338,10 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 {
 	struct graph_card_data *priv;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct graph_dai_props *dai_props;
 	struct device *dev = &pdev->dev;
 	struct snd_soc_card *card;
-	int num, ret;
+	int num, ret, i;
 
 	/* Allocate the private data and the DAI link array */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -350,6 +357,18 @@ static int asoc_graph_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->dai_props			= dai_props;
 	priv->dai_link			= dai_link;
 
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index d3f3f0fec74c..f34cc6cddfa2 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -173,12 +173,24 @@ int asoc_simple_card_parse_clk(struct device *dev,
 			       struct device_node *node,
 			       struct device_node *dai_of_node,
 			       struct asoc_simple_dai *simple_dai,
-			       const char *name)
+			       const char *dai_name,
+			       struct snd_soc_dai_link_component *dlc)
 {
 	struct clk *clk;
 	u32 val;
 
 	/*
+	 * Use snd_soc_dai_link_component instead of legacy style.
+	 * It is only for codec, but cpu will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	if (dlc) {
+		dai_of_node	= dlc->of_node;
+		dai_name	= dlc->dai_name;
+	}
+
+	/*
 	 * Parse dai->sysclk come from "clocks = <&xxx>"
 	 * (if system has common clock)
 	 *  or "system-clock-frequency = <xxx>"
@@ -200,7 +212,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
 	if (of_property_read_bool(node, "system-clock-direction-out"))
 		simple_dai->clk_direction = SND_SOC_CLOCK_OUT;
 
-	dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name,
+	dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name,
 		simple_dai->sysclk, simple_dai->clk_direction);
 
 	return 0;
@@ -208,6 +220,7 @@ int asoc_simple_card_parse_clk(struct device *dev,
 EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk);
 
 int asoc_simple_card_parse_dai(struct device_node *node,
+				    struct snd_soc_dai_link_component *dlc,
 				    struct device_node **dai_of_node,
 				    const char **dai_name,
 				    const char *list_name,
@@ -221,6 +234,17 @@ int asoc_simple_card_parse_dai(struct device_node *node,
 		return 0;
 
 	/*
+	 * Use snd_soc_dai_link_component instead of legacy style.
+	 * It is only for codec, but cpu will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	if (dlc) {
+		dai_name	= &dlc->dai_name;
+		dai_of_node	= &dlc->of_node;
+	}
+
+	/*
 	 * Get node via "sound-dai = <&phandle port>"
 	 * it will be used as xxx_of_node on soc_bind_dai_link()
 	 */
@@ -278,6 +302,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep)
 }
 
 int asoc_simple_card_parse_graph_dai(struct device_node *ep,
+				     struct snd_soc_dai_link_component *dlc,
 				     struct device_node **dai_of_node,
 				     const char **dai_name)
 {
@@ -285,6 +310,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
 	struct of_phandle_args args;
 	int ret;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style.
+	 * It is only for codec, but cpu will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	if (dlc) {
+		dai_name	= &dlc->dai_name;
+		dai_of_node	= &dlc->of_node;
+	}
+
 	if (!ep)
 		return 0;
 	if (!dai_name)
@@ -340,10 +376,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai);
 int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link)
 {
 	/* Assumes platform == cpu */
-	if (!dai_link->platform_of_node)
-		dai_link->platform_of_node = dai_link->cpu_of_node;
+	if (!dai_link->platform->of_node)
+		dai_link->platform->of_node = dai_link->cpu_of_node;
 
 	return 0;
+
 }
 EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink);
 
@@ -367,13 +404,11 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu);
 int asoc_simple_card_clean_reference(struct snd_soc_card *card)
 {
 	struct snd_soc_dai_link *dai_link;
-	int num_links;
+	int i;
 
-	for (num_links = 0, dai_link = card->dai_link;
-	     num_links < card->num_links;
-	     num_links++, dai_link++) {
+	for_each_card_prelinks(card, i, dai_link) {
 		of_node_put(dai_link->cpu_of_node);
-		of_node_put(dai_link->codec_of_node);
+		of_node_put(dai_link->codecs->of_node);
 	}
 	return 0;
 }
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 64bf3560c1d1..5a3f59aa4ba5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -20,6 +20,8 @@ struct simple_card_data {
 	struct simple_dai_props {
 		struct asoc_simple_dai cpu_dai;
 		struct asoc_simple_dai codec_dai;
+		struct snd_soc_dai_link_component codecs; /* single codec */
+		struct snd_soc_dai_link_component platform;
 		unsigned int mclk_fs;
 	} *dai_props;
 	unsigned int mclk_fs;
@@ -234,7 +236,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
 	ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 						"%s-%s",
 						dai_link->cpu_dai_name,
-						dai_link->codec_dai_name);
+						dai_link->codecs->dai_name);
 	if (ret < 0)
 		goto dai_link_of_err;
 
@@ -363,7 +365,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	struct device *dev = &pdev->dev;
 	struct device_node *np = dev->of_node;
 	struct snd_soc_card *card;
-	int num, ret;
+	int num, ret, i;
 
 	/* Get the number of DAI links */
 	if (np && of_get_child_by_name(np, PREFIX "dai-link"))
@@ -381,6 +383,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->dai_props			= dai_props;
 	priv->dai_link			= dai_link;
 
@@ -403,6 +417,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 
 	} else {
 		struct asoc_simple_card_info *cinfo;
+		struct snd_soc_dai_link_component *codecs;
+		struct snd_soc_dai_link_component *platform;
 
 		cinfo = dev->platform_data;
 		if (!cinfo) {
@@ -419,13 +435,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 			return -EINVAL;
 		}
 
+		codecs			= dai_link->codecs;
+		codecs->name		= cinfo->codec;
+		codecs->dai_name	= cinfo->codec_dai.name;
+
+		platform		= dai_link->platform;
+		platform->name		= cinfo->platform;
+
 		card->name		= (cinfo->card) ? cinfo->card : cinfo->name;
 		dai_link->name		= cinfo->name;
 		dai_link->stream_name	= cinfo->name;
-		dai_link->platform_name	= cinfo->platform;
-		dai_link->codec_name	= cinfo->codec;
 		dai_link->cpu_dai_name	= cinfo->cpu_dai.name;
-		dai_link->codec_dai_name = cinfo->codec_dai.name;
 		dai_link->dai_fmt	= cinfo->daifmt;
 		dai_link->init		= asoc_simple_card_dai_init;
 		memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai,
diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c
index 16a83bc51e0e..85b46f0eae0f 100644
--- a/sound/soc/generic/simple-scu-card.c
+++ b/sound/soc/generic/simple-scu-card.c
@@ -22,7 +22,11 @@
 struct simple_card_data {
 	struct snd_soc_card snd_card;
 	struct snd_soc_codec_conf codec_conf;
-	struct asoc_simple_dai *dai_props;
+	struct simple_dai_props {
+		struct asoc_simple_dai dai;
+		struct snd_soc_dai_link_component codecs;
+		struct snd_soc_dai_link_component platform;
+	} *dai_props;
 	struct snd_soc_dai_link *dai_link;
 	struct asoc_simple_card_data adata;
 };
@@ -40,20 +44,20 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct simple_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props =
+	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	return asoc_simple_card_clk_enable(dai_props);
+	return asoc_simple_card_clk_enable(&dai_props->dai);
 }
 
 static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct simple_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
-	struct asoc_simple_dai *dai_props =
+	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
 
-	asoc_simple_card_clk_disable(dai_props);
+	asoc_simple_card_clk_disable(&dai_props->dai);
 }
 
 static const struct snd_soc_ops asoc_simple_card_ops = {
@@ -66,7 +70,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *dai;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct simple_dai_props *dai_props;
 	int num = rtd->num;
 
 	dai_link	= simple_priv_to_link(priv, num);
@@ -75,7 +79,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 				rtd->cpu_dai :
 				rtd->codec_dai;
 
-	return asoc_simple_card_init_dai(dai, dai_props);
+	return asoc_simple_card_init_dai(dai, &dai_props->dai);
 }
 
 static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -95,17 +99,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 {
 	struct device *dev = simple_priv_to_dev(priv);
 	struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx);
-	struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx);
+	struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx);
 	struct snd_soc_card *card = simple_priv_to_card(priv);
 	int ret;
 
 	if (is_fe) {
 		int is_single_links = 0;
+		struct snd_soc_dai_link_component *codecs;
 
 		/* BE is dummy */
-		dai_link->codec_of_node		= NULL;
-		dai_link->codec_dai_name	= "snd-soc-dummy-dai";
-		dai_link->codec_name		= "snd-soc-dummy";
+		codecs			= dai_link->codecs;
+		codecs->of_node		= NULL;
+		codecs->dai_name	= "snd-soc-dummy-dai";
+		codecs->name		= "snd-soc-dummy";
 
 		/* FE settings */
 		dai_link->dynamic		= 1;
@@ -116,7 +122,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 		if (ret)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
@@ -141,23 +147,23 @@ static int asoc_simple_card_dai_link_of(struct device_node *np,
 		if (ret < 0)
 			return ret;
 
-		ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props);
+		ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai);
 		if (ret < 0)
 			return ret;
 
 		ret = asoc_simple_card_set_dailink_name(dev, dai_link,
 							"be.%s",
-							dai_link->codec_dai_name);
+							dai_link->codecs->dai_name);
 		if (ret < 0)
 			return ret;
 
 		snd_soc_of_parse_audio_prefix(card,
 					      &priv->codec_conf,
-					      dai_link->codec_of_node,
+					      dai_link->codecs->of_node,
 					      PREFIX "prefix");
 	}
 
-	ret = asoc_simple_card_of_parse_tdm(np, dai_props);
+	ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai);
 	if (ret)
 		return ret;
 
@@ -230,11 +236,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 {
 	struct simple_card_data *priv;
 	struct snd_soc_dai_link *dai_link;
-	struct asoc_simple_dai *dai_props;
+	struct simple_dai_props *dai_props;
 	struct snd_soc_card *card;
 	struct device *dev = &pdev->dev;
 	struct device_node *np = dev->of_node;
-	int num, ret;
+	int num, ret, i;
 
 	/* Allocate the private data */
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -248,6 +254,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	if (!dai_props || !dai_link)
 		return -ENOMEM;
 
+	/*
+	 * Use snd_soc_dai_link_component instead of legacy style
+	 * It is codec only. but cpu/platform will be supported in the future.
+	 * see
+	 *	soc-core.c :: snd_soc_init_multicodec()
+	 */
+	for (i = 0; i < num; i++) {
+		dai_link[i].codecs	= &dai_props[i].codecs;
+		dai_link[i].num_codecs	= 1;
+		dai_link[i].platform	= &dai_props[i].platform;
+	}
+
 	priv->dai_props				= dai_props;
 	priv->dai_link				= dai_link;
 
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index 53344a3b7a60..a69e5b11b3da 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U16_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bits = HII2S_BITS_16;
 		break;
 	case SNDRV_PCM_FORMAT_U24_LE:
 		signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT;
-		/* fallthru */
+		/* fall through */
 	case SNDRV_PCM_FORMAT_S24_LE:
 		bits = HII2S_BITS_24;
 		break;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 6c36da560877..afc559866095 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -765,7 +765,7 @@ static int sst_soc_prepare(struct device *dev)
 	snd_soc_poweroff(drv->soc_card->dev);
 
 	/* set the SSPs to idle */
-	list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+	for_each_card_rtds(drv->soc_card, rtd) {
 		struct snd_soc_dai *dai = rtd->cpu_dai;
 
 		if (dai->active) {
@@ -786,7 +786,7 @@ static void sst_soc_complete(struct device *dev)
 		return;
 
 	/* restart SSPs */
-	list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) {
+	for_each_card_rtds(drv->soc_card, rtd) {
 		struct snd_soc_dai *dai = rtd->cpu_dai;
 
 		if (dai->active) {
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index cccda87f4b34..73ca1350aa31 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -279,6 +279,28 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH
 	  This adds support for ASoC Onboard Codec I2S machine driver. This will
 	  create an alsa sound card for DA7219 + MAX98357A I2S audio codec.
 	  Say Y if you have such a device.
+
+config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
+	tristate "KBL with DA7219 and MAX98927 in I2S Mode"
+	depends on MFD_INTEL_LPSS && I2C && ACPI
+	select SND_SOC_DA7219
+	select SND_SOC_MAX98927
+	select SND_SOC_DMIC
+	select SND_SOC_HDAC_HDMI
+	help
+	  This adds support for ASoC Onboard Codec I2S machine driver. This will
+	  create an alsa sound card for DA7219 + MAX98927 I2S audio codec.
+	  Say Y if you have such a device.
+	  If unsure select "N".
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+	tristate "SKL/KBL/BXT/APL with HDA Codecs"
+	select SND_SOC_HDAC_HDMI
+	select SND_SOC_HDAC_HDA
+	help
+	  This adds support for ASoC machine driver for Intel platforms
+	  SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+          Say Y or m if you have such a device. This is a recommended option.
 	  If unsure select "N".
 
 config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
index 87ef8b4058e5..5381e27df9cc 100644
--- a/sound/soc/intel/boards/Makefile
+++ b/sound/soc/intel/boards/Makefile
@@ -17,9 +17,11 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
 snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
 snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
 snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
+snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
 snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o
 snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o
 snd-soc-skl_rt286-objs := skl_rt286.o
+snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o
 snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
 snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
 
@@ -41,8 +43,10 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o
+obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
+obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 7b0ee67b4fc8..68e6543e6cb0 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 static int broadwell_suspend(struct snd_soc_card *card){
 	struct snd_soc_component *component;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, "i2c-INT343A:00")) {
 
 			dev_dbg(component->dev, "disabling jack detect before going to suspend.\n");
@@ -237,7 +237,7 @@ static int broadwell_suspend(struct snd_soc_card *card){
 static int broadwell_resume(struct snd_soc_card *card){
 	struct snd_soc_component *component;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, "i2c-INT343A:00")) {
 
 			dev_dbg(component->dev, "enabling jack detect for resume.\n");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index b6dc524830b2..8587bd3d1cc1 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -1048,7 +1048,7 @@ static int byt_rt5640_suspend(struct snd_soc_card *card)
 	if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5640_codec_name)) {
 			dev_dbg(component->dev, "disabling jack detect before suspend\n");
 			snd_soc_component_set_jack(component, NULL, NULL);
@@ -1067,7 +1067,7 @@ static int byt_rt5640_resume(struct snd_soc_card *card)
 	if (!BYT_RT5640_JDSRC(byt_rt5640_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5640_codec_name)) {
 			dev_dbg(component->dev, "re-enabling jack detect after resume\n");
 			snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index f8a68bdb3885..8dffeecda55b 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -742,7 +742,7 @@ static int byt_rt5651_suspend(struct snd_soc_card *card)
 	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5651_codec_name)) {
 			dev_dbg(component->dev, "disabling jack detect before suspend\n");
 			snd_soc_component_set_jack(component, NULL, NULL);
@@ -761,7 +761,7 @@ static int byt_rt5651_resume(struct snd_soc_card *card)
 	if (!BYT_RT5651_JDSRC(byt_rt5651_quirk))
 		return 0;
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strcmp(component->name, byt_rt5651_codec_name)) {
 			dev_dbg(component->dev, "re-enabling jack detect after resume\n");
 			snd_soc_component_set_jack(component, &priv->jack, NULL);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index e5aa13058dd7..51f0d45d6f8f 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -16,6 +16,7 @@
  *  General Public License for more details.
  */
 
+#include <linux/input.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -212,6 +213,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
         if (ret)
                 return ret;
 
+	snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(ctx->headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+
 	rt5670_set_jack_detect(component, &ctx->headset);
 	if (ctx->mclk) {
 		/*
@@ -342,7 +347,7 @@ static int cht_suspend_pre(struct snd_soc_card *card)
 	struct snd_soc_component *component;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strncmp(component->name,
 			     ctx->codec_name, sizeof(ctx->codec_name))) {
 
@@ -359,7 +364,7 @@ static int cht_resume_post(struct snd_soc_card *card)
 	struct snd_soc_component *component;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (!strncmp(component->name,
 			     ctx->codec_name, sizeof(ctx->codec_name))) {
 
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
new file mode 100644
index 000000000000..3fa1c3ca6d37
--- /dev/null
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -0,0 +1,983 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018 Intel Corporation.
+
+/*
+ * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs
+ *
+ * Modified from:
+ *   Intel Kabylake I2S Machine driver supporting MAX98927 and
+ *   RT5663 codecs
+ */
+
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/da7219.h"
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "../../codecs/da7219-aad.h"
+
+#define KBL_DIALOG_CODEC_DAI	"da7219-hifi"
+#define MAX98927_CODEC_DAI	"max98927-aif1"
+#define MAXIM_DEV0_NAME		"i2c-MX98927:00"
+#define MAXIM_DEV1_NAME		"i2c-MX98927:01"
+#define DUAL_CHANNEL	2
+#define QUAD_CHANNEL	4
+#define NAME_SIZE	32
+
+static struct snd_soc_card *kabylake_audio_card;
+static struct snd_soc_jack kabylake_hdmi[3];
+
+struct kbl_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct kbl_codec_private {
+	struct snd_soc_jack kabylake_headset;
+	struct list_head hdmi_pcm_list;
+};
+
+enum {
+	KBL_DPCM_AUDIO_PB = 0,
+	KBL_DPCM_AUDIO_CP,
+	KBL_DPCM_AUDIO_ECHO_REF_CP,
+	KBL_DPCM_AUDIO_REF_CP,
+	KBL_DPCM_AUDIO_DMIC_CP,
+	KBL_DPCM_AUDIO_HDMI1_PB,
+	KBL_DPCM_AUDIO_HDMI2_PB,
+	KBL_DPCM_AUDIO_HDMI3_PB,
+	KBL_DPCM_AUDIO_HS_PB,
+};
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+					struct snd_kcontrol *k, int  event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct snd_soc_dai *codec_dai;
+	int ret = 0;
+
+	codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI);
+	if (!codec_dai) {
+		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
+		return -EIO;
+	}
+
+	/* Configure sysclk for codec */
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000,
+				     SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(card->dev, "can't set codec sysclk configuration\n");
+		return ret;
+	}
+
+	if (SND_SOC_DAPM_EVENT_OFF(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0,
+				     DA7219_SYSCLK_MCLK, 0, 0);
+		if (ret)
+			dev_err(card->dev, "failed to stop PLL: %d\n", ret);
+	} else if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0,	DA7219_SYSCLK_PLL_SRM,
+				     0, DA7219_PLL_FREQ_OUT_98304);
+		if (ret)
+			dev_err(card->dev, "failed to start PLL: %d\n", ret);
+	}
+
+	return ret;
+}
+
+static const struct snd_kcontrol_new kabylake_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const struct snd_soc_dapm_widget kabylake_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+	SND_SOC_DAPM_SPK("DP", NULL),
+	SND_SOC_DAPM_SPK("HDMI", NULL),
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route kabylake_map[] = {
+	/* speaker */
+	{ "Left Spk", NULL, "Left BE_OUT" },
+	{ "Right Spk", NULL, "Right BE_OUT" },
+
+	/* other jacks */
+	{ "DMic", NULL, "SoC DMIC" },
+
+	{ "HDMI", NULL, "hif5 Output" },
+	{ "DP", NULL, "hif6 Output" },
+
+	/* CODEC BE connections */
+	{ "Left HiFi Playback", NULL, "ssp0 Tx" },
+	{ "Right HiFi Playback", NULL, "ssp0 Tx" },
+	{ "ssp0 Tx", NULL, "spk_out" },
+
+	/* IV feedback path */
+	{ "codec0_fb_in", NULL, "ssp0 Rx"},
+	{ "ssp0 Rx", NULL, "Left HiFi Capture" },
+	{ "ssp0 Rx", NULL, "Right HiFi Capture" },
+
+	/* AEC capture path */
+	{ "echo_ref_out", NULL, "ssp0 Rx" },
+
+	/* DMIC */
+	{ "dmic01_hifi", NULL, "DMIC01 Rx" },
+	{ "DMIC01 Rx", NULL, "DMIC AIF" },
+
+	{ "hifi1", NULL, "iDisp1 Tx" },
+	{ "iDisp1 Tx", NULL, "iDisp1_out" },
+	{ "hifi2", NULL, "iDisp2 Tx" },
+	{ "iDisp2 Tx", NULL, "iDisp2_out" },
+	{ "hifi3", NULL, "iDisp3 Tx"},
+	{ "iDisp3 Tx", NULL, "iDisp3_out"},
+};
+
+static const struct snd_soc_dapm_route kabylake_ssp1_map[] = {
+	{ "Headphone Jack", NULL, "HPL" },
+	{ "Headphone Jack", NULL, "HPR" },
+
+	/* other jacks */
+	{ "MIC", NULL, "Headset Mic" },
+
+	/* CODEC BE connections */
+	{ "Playback", NULL, "ssp1 Tx" },
+	{ "ssp1 Tx", NULL, "codec1_out" },
+
+	{ "hs_in", NULL, "ssp1 Rx" },
+	{ "ssp1 Rx", NULL, "Capture" },
+
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headset Mic", NULL, "Platform Clock" },
+};
+
+static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *runtime = substream->private_data;
+	int ret = 0, j;
+
+	for (j = 0; j < runtime->num_codecs; j++) {
+		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+		if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
+			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+			if (ret < 0) {
+				dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
+				return ret;
+			}
+		}
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops kabylake_ssp0_ops = {
+	.hw_params = kabylake_ssp0_hw_params,
+};
+
+static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_soc_dpcm *dpcm = container_of(
+			params, struct snd_soc_dpcm, hw_params);
+	struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
+	struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+
+	/*
+	 * The ADSP will convert the FE rate to 48k, stereo, 24 bit
+	 */
+	if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
+	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
+	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+		rate->min = rate->max = 48000;
+		channels->min = channels->max = 2;
+		snd_mask_none(fmt);
+		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	}
+
+	/*
+	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
+	 * thus changing the mask here
+	 */
+	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+		snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE);
+
+	return 0;
+}
+
+static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_jack *jack;
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+
+	ret = snd_soc_dapm_add_routes(&card->dapm,
+			kabylake_ssp1_map,
+			ARRAY_SIZE(kabylake_ssp1_map));
+
+	/*
+	 * Headset buttons map to the google Reference headset.
+	 * These can be configured by userspace.
+	 */
+	ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT,
+			&ctx->kabylake_headset, NULL, 0);
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
+		return ret;
+	}
+
+	jack = &ctx->kabylake_headset;
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
+	da7219_aad_jack_det(component, &ctx->kabylake_headset);
+
+	ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC");
+	if (ret)
+		dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret);
+
+	return ret;
+}
+
+static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
+{
+	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct kbl_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	pcm->device = device;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB);
+}
+
+static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB);
+}
+
+static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
+{
+	return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB);
+}
+
+static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dapm_context *dapm;
+	struct snd_soc_component *component = rtd->cpu_dai->component;
+
+	dapm = snd_soc_component_get_dapm(component);
+	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
+
+	return 0;
+}
+
+static const unsigned int rates[] = {
+	48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+	.count = ARRAY_SIZE(rates),
+	.list  = rates,
+	.mask = 0,
+};
+
+static const unsigned int channels[] = {
+	DUAL_CHANNEL,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+	.count = ARRAY_SIZE(channels),
+	.list = channels,
+	.mask = 0,
+};
+
+static unsigned int channels_quad[] = {
+	QUAD_CHANNEL,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
+	.count = ARRAY_SIZE(channels_quad),
+	.list = channels_quad,
+	.mask = 0,
+};
+
+static int kbl_fe_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	/*
+	 * On this platform for PCM device we support,
+	 * 48Khz
+	 * stereo
+	 * 16 bit audio
+	 */
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+					   &constraints_channels);
+
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+	snd_pcm_hw_constraint_list(runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+
+	return 0;
+}
+
+static const struct snd_soc_ops kabylake_da7219_fe_ops = {
+	.startup = kbl_fe_startup,
+};
+
+static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
+		struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *channels = hw_param_interval(params,
+				SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/*
+	 * set BE channel constraint as user FE channels
+	 */
+
+	if (params_channels(params) == 2)
+		channels->min = channels->max = 2;
+	else
+		channels->min = channels->max = 4;
+
+	return 0;
+}
+
+static int kabylake_dmic_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+			&constraints_channels_quad);
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+}
+
+static struct snd_soc_ops kabylake_dmic_ops = {
+	.startup = kabylake_dmic_startup,
+};
+
+static const unsigned int rates_16000[] = {
+	16000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_16000 = {
+	.count = ARRAY_SIZE(rates_16000),
+	.list  = rates_16000,
+};
+
+static const unsigned int ch_mono[] = {
+	1,
+};
+static const struct snd_pcm_hw_constraint_list constraints_refcap = {
+	.count = ARRAY_SIZE(ch_mono),
+	.list  = ch_mono,
+};
+
+static int kabylake_refcap_startup(struct snd_pcm_substream *substream)
+{
+	substream->runtime->hw.channels_max = 1;
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+					SNDRV_PCM_HW_PARAM_CHANNELS,
+					&constraints_refcap);
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				&constraints_16000);
+}
+
+
+static struct snd_soc_ops skylaye_refcap_ops = {
+	.startup = kabylake_refcap_startup,
+};
+
+static struct snd_soc_codec_conf max98927_codec_conf[] = {
+
+	{
+		.dev_name = MAXIM_DEV0_NAME,
+		.name_prefix = "Right",
+	},
+
+	{
+		.dev_name = MAXIM_DEV1_NAME,
+		.name_prefix = "Left",
+	},
+};
+
+static struct snd_soc_dai_link_component ssp0_codec_components[] = {
+	{ /* Left */
+		.name = MAXIM_DEV0_NAME,
+		.dai_name = MAX98927_CODEC_DAI,
+	},
+
+	{  /* For Right */
+		.name = MAXIM_DEV1_NAME,
+		.dai_name = MAX98927_CODEC_DAI,
+	},
+
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_dais[] = {
+	/* Front End DAI links */
+	[KBL_DPCM_AUDIO_PB] = {
+		.name = "Kbl Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.init = kabylake_da7219_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_CP] = {
+		.name = "Kbl Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Kbl Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
+	[KBL_DPCM_AUDIO_REF_CP] = {
+		.name = "Kbl Audio Reference cap",
+		.stream_name = "Wake on Voice",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &skylaye_refcap_ops,
+	},
+	[KBL_DPCM_AUDIO_DMIC_CP] = {
+		.name = "Kbl Audio DMIC cap",
+		.stream_name = "dmiccap",
+		.cpu_dai_name = "DMIC Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &kabylake_dmic_ops,
+	},
+	[KBL_DPCM_AUDIO_HDMI1_PB] = {
+		.name = "Kbl HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI2_PB] = {
+		.name = "Kbl HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI3_PB] = {
+		.name = "Kbl HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HS_PB] = {
+		.name = "Kbl Audio Headset Playback",
+		.stream_name = "Headset Audio",
+		.cpu_dai_name = "System Pin2",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.init = kabylake_da7219_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.ops = &kabylake_da7219_fe_ops,
+
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "SSP0-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP0 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codecs = ssp0_codec_components,
+		.num_codecs = ARRAY_SIZE(ssp0_codec_components),
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.ops = &kabylake_ssp0_ops,
+	},
+	{
+		/* SSP1 - Codec */
+		.name = "SSP1-Codec",
+		.id = 1,
+		.cpu_dai_name = "SSP1 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codec_name = "i2c-DLGS7219:00",
+		.codec_dai_name = KBL_DIALOG_CODEC_DAI,
+		.init = kabylake_da7219_codec_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	{
+		.name = "dmic01",
+		.id = 2,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:1f.3",
+		.be_hw_params_fixup = kabylake_dmic_fixup,
+		.ignore_suspend = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 3,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = kabylake_hdmi1_init,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 4,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi2_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 5,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi3_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+/* kabylake digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link kabylake_max98927_dais[] = {
+	/* Front End DAI links */
+	[KBL_DPCM_AUDIO_PB] = {
+		.name = "Kbl Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.init = kabylake_da7219_fe_init,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_CP] = {
+		.name = "Kbl Audio Capture Port",
+		.stream_name = "Audio Record",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "0000:00:1f.3",
+		.dynamic = 1,
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.nonatomic = 1,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_capture = 1,
+		.ops = &kabylake_da7219_fe_ops,
+	},
+	[KBL_DPCM_AUDIO_ECHO_REF_CP] = {
+		.name = "Kbl Audio Echo Reference cap",
+		.stream_name = "Echoreference Capture",
+		.cpu_dai_name = "Echoref Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.capture_only = 1,
+		.nonatomic = 1,
+	},
+	[KBL_DPCM_AUDIO_REF_CP] = {
+		.name = "Kbl Audio Reference cap",
+		.stream_name = "Wake on Voice",
+		.cpu_dai_name = "Reference Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &skylaye_refcap_ops,
+	},
+	[KBL_DPCM_AUDIO_DMIC_CP] = {
+		.name = "Kbl Audio DMIC cap",
+		.stream_name = "dmiccap",
+		.cpu_dai_name = "DMIC Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.init = NULL,
+		.dpcm_capture = 1,
+		.nonatomic = 1,
+		.dynamic = 1,
+		.ops = &kabylake_dmic_ops,
+	},
+	[KBL_DPCM_AUDIO_HDMI1_PB] = {
+		.name = "Kbl HDMI Port1",
+		.stream_name = "Hdmi1",
+		.cpu_dai_name = "HDMI1 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI2_PB] = {
+		.name = "Kbl HDMI Port2",
+		.stream_name = "Hdmi2",
+		.cpu_dai_name = "HDMI2 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = NULL,
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+	[KBL_DPCM_AUDIO_HDMI3_PB] = {
+		.name = "Kbl HDMI Port3",
+		.stream_name = "Hdmi3",
+		.cpu_dai_name = "HDMI3 Pin",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.platform_name = "0000:00:1f.3",
+		.trigger = {
+			SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+		.init = NULL,
+		.nonatomic = 1,
+		.dynamic = 1,
+	},
+
+	/* Back End DAI links */
+	{
+		/* SSP0 - Codec */
+		.name = "SSP0-Codec",
+		.id = 0,
+		.cpu_dai_name = "SSP0 Pin",
+		.platform_name = "0000:00:1f.3",
+		.no_pcm = 1,
+		.codecs = ssp0_codec_components,
+		.num_codecs = ARRAY_SIZE(ssp0_codec_components),
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = kabylake_ssp_fixup,
+		.ops = &kabylake_ssp0_ops,
+	},
+	{
+		.name = "dmic01",
+		.id = 1,
+		.cpu_dai_name = "DMIC01 Pin",
+		.codec_name = "dmic-codec",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "0000:00:1f.3",
+		.be_hw_params_fixup = kabylake_dmic_fixup,
+		.ignore_suspend = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp1",
+		.id = 2,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.init = kabylake_hdmi1_init,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 3,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi2_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 4,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.platform_name = "0000:00:1f.3",
+		.init = kabylake_hdmi3_init,
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int kabylake_card_late_probe(struct snd_soc_card *card)
+{
+	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card);
+	struct kbl_hdmi_pcm *pcm;
+	struct snd_soc_component *component = NULL;
+	int err, i = 0;
+	char jack_name[NAME_SIZE];
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			"HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					SND_JACK_AVOUT, &kabylake_hdmi[i],
+					NULL, 0);
+
+		if (err)
+			return err;
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+						&kabylake_hdmi[i]);
+		if (err < 0)
+			return err;
+
+		i++;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+
+	return 0;
+}
+
+/* kabylake audio machine driver for SPT + DA7219 */
+static struct snd_soc_card kbl_audio_card_da7219_m98927 = {
+	.name = "kblda7219m98927",
+	.owner = THIS_MODULE,
+	.dai_link = kabylake_dais,
+	.num_links = ARRAY_SIZE(kabylake_dais),
+	.controls = kabylake_controls,
+	.num_controls = ARRAY_SIZE(kabylake_controls),
+	.dapm_widgets = kabylake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+	.dapm_routes = kabylake_map,
+	.num_dapm_routes = ARRAY_SIZE(kabylake_map),
+	.codec_conf = max98927_codec_conf,
+	.num_configs = ARRAY_SIZE(max98927_codec_conf),
+	.fully_routed = true,
+	.late_probe = kabylake_card_late_probe,
+};
+
+/* kabylake audio machine driver for Maxim98927 */
+static struct snd_soc_card kbl_audio_card_max98927 = {
+	.name = "kblmax98927",
+	.owner = THIS_MODULE,
+	.dai_link = kabylake_max98927_dais,
+	.num_links = ARRAY_SIZE(kabylake_max98927_dais),
+	.controls = kabylake_controls,
+	.num_controls = ARRAY_SIZE(kabylake_controls),
+	.dapm_widgets = kabylake_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(kabylake_widgets),
+	.dapm_routes = kabylake_map,
+	.num_dapm_routes = ARRAY_SIZE(kabylake_map),
+	.codec_conf = max98927_codec_conf,
+	.num_configs = ARRAY_SIZE(max98927_codec_conf),
+	.fully_routed = true,
+	.late_probe = kabylake_card_late_probe,
+};
+
+static int kabylake_audio_probe(struct platform_device *pdev)
+{
+	struct kbl_codec_private *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	kabylake_audio_card =
+		(struct snd_soc_card *)pdev->id_entry->driver_data;
+
+	kabylake_audio_card->dev = &pdev->dev;
+	snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card);
+}
+
+static const struct platform_device_id kbl_board_ids[] = {
+	{
+		.name = "kbl_da7219_max98927",
+		.driver_data =
+			(kernel_ulong_t)&kbl_audio_card_da7219_m98927,
+	},
+	{
+		.name = "kbl_max98927",
+		.driver_data =
+			(kernel_ulong_t)&kbl_audio_card_max98927,
+	},
+	{ }
+};
+
+static struct platform_driver kabylake_audio = {
+	.probe = kabylake_audio_probe,
+	.driver = {
+		.name = "kbl_da7219_max98927",
+		.pm = &snd_soc_pm_ops,
+	},
+	.id_table = kbl_board_ids,
+};
+
+module_platform_driver(kabylake_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219");
+MODULE_AUTHOR("Mac Chiang <mac.chiang@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:kbl_da7219_max98927");
+MODULE_ALIAS("platform:kbl_max98927");
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 21a6490746a6..99e1320c485f 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -488,11 +488,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int ret = 0, j;
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+	for_each_rtd_codec_dai(rtd, j, codec_dai) {
 		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
 			/*
 			 * Use channel 4 and 5 for the first amp
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index a892b37eab7c..a737c915d46a 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -353,11 +353,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int ret = 0, j;
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-
+	for_each_rtd_codec_dai(rtd, j, codec_dai) {
 		if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) {
 			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16);
 			if (ret < 0) {
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c
new file mode 100644
index 000000000000..3fdbf239da74
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.c
@@ -0,0 +1,127 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Common functions used in different Intel machine drivers
+ */
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+#define NAME_SIZE	32
+
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device)
+{
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+	struct skl_hda_hdmi_pcm *pcm;
+	char dai_name[NAME_SIZE];
+
+	pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d",
+		 ctx->dai_index);
+	pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name);
+	if (!pcm->codec_dai)
+		return -EINVAL;
+
+	pcm->device = device;
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+/* skl_hda_digital audio interface glue - connects codec <--> CPU */
+struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = {
+	/* Back End DAI links */
+	{
+		.name = "iDisp1",
+		.id = 1,
+		.cpu_dai_name = "iDisp1 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi1",
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp2",
+		.id = 2,
+		.cpu_dai_name = "iDisp2 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi2",
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "iDisp3",
+		.id = 3,
+		.cpu_dai_name = "iDisp3 Pin",
+		.codec_name = "ehdaudio0D2",
+		.codec_dai_name = "intel-hdmi-hifi3",
+		.dpcm_playback = 1,
+		.no_pcm = 1,
+	},
+	{
+		.name = "Analog Playback and Capture",
+		.id = 4,
+		.cpu_dai_name = "Analog CPU DAI",
+		.codec_name = "ehdaudio0D0",
+		.codec_dai_name = "Analog Codec DAI",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.init = NULL,
+		.no_pcm = 1,
+	},
+	{
+		.name = "Digital Playback and Capture",
+		.id = 5,
+		.cpu_dai_name = "Digital CPU DAI",
+		.codec_name = "ehdaudio0D0",
+		.codec_dai_name = "Digital Codec DAI",
+		.platform_name = "0000:00:1f.3",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.init = NULL,
+		.no_pcm = 1,
+	},
+};
+
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card)
+{
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = NULL;
+	struct skl_hda_hdmi_pcm *pcm;
+	char jack_name[NAME_SIZE];
+	int err;
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			 "HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					    SND_JACK_AVOUT, &pcm->hdmi_jack,
+					    NULL, 0);
+
+		if (err)
+			return err;
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+					  &pcm->hdmi_jack);
+		if (err < 0)
+			return err;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h
new file mode 100644
index 000000000000..87c50aff56cd
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.h
@@ -0,0 +1,38 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2015-18 Intel Corporation.
+ */
+
+/*
+ * This file defines data structures used in Machine Driver for Intel
+ * platforms with HDA Codecs.
+ */
+
+#ifndef __SOUND_SOC_HDA_DSP_COMMON_H
+#define __SOUND_SOC_HDA_DSP_COMMON_H
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+
+#define HDA_DSP_MAX_BE_DAI_LINKS 5
+
+struct skl_hda_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_jack hdmi_jack;
+	int device;
+};
+
+struct skl_hda_private {
+	struct list_head hdmi_pcm_list;
+	int pcm_count;
+	int dai_index;
+	const char *platform_name;
+};
+
+extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS];
+int skl_hda_hdmi_jack_init(struct snd_soc_card *card);
+int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device);
+
+#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
new file mode 100644
index 000000000000..b415dd4c85f5
--- /dev/null
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -0,0 +1,183 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2015-18 Intel Corporation.
+
+/*
+ * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/hdac_hdmi.h"
+#include "../skylake/skl.h"
+#include "skl_hda_dsp_common.h"
+
+static const struct snd_soc_dapm_widget skl_hda_widgets[] = {
+	SND_SOC_DAPM_HP("Analog Out", NULL),
+	SND_SOC_DAPM_MIC("Analog In", NULL),
+	SND_SOC_DAPM_HP("Alt Analog Out", NULL),
+	SND_SOC_DAPM_MIC("Alt Analog In", NULL),
+	SND_SOC_DAPM_SPK("Digital Out", NULL),
+	SND_SOC_DAPM_MIC("Digital In", NULL),
+};
+
+static const struct snd_soc_dapm_route skl_hda_map[] = {
+	{ "hifi3", NULL, "iDisp3 Tx"},
+	{ "iDisp3 Tx", NULL, "iDisp3_out"},
+	{ "hifi2", NULL, "iDisp2 Tx"},
+	{ "iDisp2 Tx", NULL, "iDisp2_out"},
+	{ "hifi1", NULL, "iDisp1 Tx"},
+	{ "iDisp1 Tx", NULL, "iDisp1_out"},
+
+	{ "Analog Out", NULL, "Codec Output Pin1" },
+	{ "Digital Out", NULL, "Codec Output Pin2" },
+	{ "Alt Analog Out", NULL, "Codec Output Pin3" },
+
+	{ "Codec Input Pin1", NULL, "Analog In" },
+	{ "Codec Input Pin2", NULL, "Digital In" },
+	{ "Codec Input Pin3", NULL, "Alt Analog In" },
+
+	/* CODEC BE connections */
+	{ "Analog Codec Playback", NULL, "Analog CPU Playback" },
+	{ "Analog CPU Playback", NULL, "codec0_out" },
+	{ "Digital Codec Playback", NULL, "Digital CPU Playback" },
+	{ "Digital CPU Playback", NULL, "codec1_out" },
+	{ "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" },
+	{ "Alt Analog CPU Playback", NULL, "codec2_out" },
+
+	{ "codec0_in", NULL, "Analog CPU Capture" },
+	{ "Analog CPU Capture", NULL, "Analog Codec Capture" },
+	{ "codec1_in", NULL, "Digital CPU Capture" },
+	{ "Digital CPU Capture", NULL, "Digital Codec Capture" },
+	{ "codec2_in", NULL, "Alt Analog CPU Capture" },
+	{ "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" },
+};
+
+static int skl_hda_card_late_probe(struct snd_soc_card *card)
+{
+	return skl_hda_hdmi_jack_init(card);
+}
+
+static int
+skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
+{
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
+	int ret = 0;
+
+	dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name);
+	link->platform_name = ctx->platform_name;
+	link->nonatomic = 1;
+
+	if (strstr(link->name, "HDMI")) {
+		ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count);
+
+		if (ret < 0)
+			return ret;
+
+		ctx->dai_index++;
+	}
+
+	ctx->pcm_count++;
+	return ret;
+}
+
+static struct snd_soc_card hda_soc_card = {
+	.name = "skl_hda_card",
+	.owner = THIS_MODULE,
+	.dai_link = skl_hda_be_dai_links,
+	.dapm_widgets = skl_hda_widgets,
+	.dapm_routes = skl_hda_map,
+	.add_dai_link = skl_hda_add_dai_link,
+	.fully_routed = true,
+	.late_probe = skl_hda_card_late_probe,
+};
+
+#define IDISP_DAI_COUNT		3
+/* there are two routes per iDisp output */
+#define IDISP_ROUTE_COUNT	(IDISP_DAI_COUNT * 2)
+#define IDISP_CODEC_MASK	0x4
+
+static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata)
+{
+	struct snd_soc_card *card = &hda_soc_card;
+	struct snd_soc_dai_link *dai_link;
+	u32 codec_count, codec_mask;
+	int i, num_links, num_route;
+
+	codec_mask = pdata->codec_mask;
+	codec_count = hweight_long(codec_mask);
+
+	if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) {
+		num_links = IDISP_DAI_COUNT;
+		num_route = IDISP_ROUTE_COUNT;
+	} else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) {
+		num_links = ARRAY_SIZE(skl_hda_be_dai_links);
+		num_route = ARRAY_SIZE(skl_hda_map),
+		card->dapm_widgets = skl_hda_widgets;
+		card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
+	} else {
+		return -EINVAL;
+	}
+
+	card->num_links = num_links;
+	card->num_dapm_routes = num_route;
+
+	for_each_card_prelinks(card, i, dai_link)
+		dai_link->platform_name = pdata->platform;
+
+	return 0;
+}
+
+static int skl_hda_audio_probe(struct platform_device *pdev)
+{
+	struct skl_machine_pdata *pdata;
+	struct skl_hda_private *ctx;
+	int ret;
+
+	dev_dbg(&pdev->dev, "%s: entry\n", __func__);
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	if (!ctx)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	pdata = dev_get_drvdata(&pdev->dev);
+	if (!pdata)
+		return -EINVAL;
+
+	ret = skl_hda_fill_card_info(pdata);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
+		return ret;
+	}
+
+	ctx->pcm_count = hda_soc_card.num_links;
+	ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */
+	ctx->platform_name = pdata->platform;
+
+	hda_soc_card.dev = &pdev->dev;
+	snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
+}
+
+static struct platform_driver skl_hda_audio = {
+	.probe = skl_hda_audio_probe,
+	.driver = {
+		.name = "skl_hda_dsp_generic",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(skl_hda_audio)
+
+/* Module information */
+MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver");
+MODULE_AUTHOR("Rakesh Ughreja <rakesh.a.ughreja@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:skl_hda_dsp_generic");
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 915a34cdc8ac..c1f50a079d34 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m
 	soc-acpi-intel-hsw-bdw-match.o \
 	soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \
 	soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \
-	soc-acpi-intel-cnl-match.o
+	soc-acpi-intel-cnl-match.o \
+	soc-acpi-intel-hda-match.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
 obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index 4daa8a4f0c0c..097dc06377ba 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -34,6 +34,13 @@ static const struct dmi_system_id byt_table[] = {
 		.callback = byt_thinkpad10_quirk_cb,
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"),
+		},
+	},
+	{
+		.callback = byt_thinkpad10_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 10"),
 		},
 	},
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
new file mode 100644
index 000000000000..533c1064f84b
--- /dev/null
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -0,0 +1,40 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Intel Corporation.
+
+/*
+ * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ *
+ */
+
+#include <sound/soc-acpi.h>
+#include <sound/soc-acpi-intel-match.h>
+#include "../skylake/skl.h"
+
+static struct skl_machine_pdata hda_pdata = {
+	.use_tplg_pcm = true,
+};
+
+struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = {
+	{
+		/* .id is not used in this file */
+		.drv_name = "skl_hda_dsp_generic",
+
+		/* .fw_filename is dynamically set in skylake driver */
+
+		/* .sof_fw_filename is dynamically set in sof/intel driver */
+
+		.sof_tplg_filename = "intel/sof-hda-generic.tplg",
+
+		/*
+		 * .machine_quirk and .quirk_data are not used here but
+		 * can be used if we need a more complicated machine driver
+		 * combining HDA+other device (e.g. DMIC).
+		 */
+		.pdata = &hda_pdata,
+	},
+	{},
+};
+EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 0ee173ca437d..a317b7790fce 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = {
 	.codecs = {"MX98357A"}
 };
 
+static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98927"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
 	{
 		.id = "INT343A",
@@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = {
 		.quirk_data = &kbl_7219_98357_codecs,
 		.pdata = &skl_dmic_data,
 	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "kbl_da7219_max98927",
+		.fw_filename = "intel/dsp_fw_kbl.bin",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &kbl_7219_98927_codecs,
+		.pdata = &skl_dmic_data
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines);
diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c
index 11041aedea31..1e067504b604 100644
--- a/sound/soc/intel/common/sst-firmware.c
+++ b/sound/soc/intel/common/sst-firmware.c
@@ -355,7 +355,7 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
 
 	/* allocate DMA buffer to store FW data */
 	sst_fw->dma_buf = dma_alloc_coherent(dsp->dma_dev, sst_fw->size,
-				&sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
+				&sst_fw->dmable_fw_paddr, GFP_KERNEL);
 	if (!sst_fw->dma_buf) {
 		dev_err(dsp->dev, "error: DMA alloc failed\n");
 		kfree(sst_fw);
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 823e39103edd..557f80c0bfe5 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -32,6 +32,7 @@
 #define HDA_MONO 1
 #define HDA_STEREO 2
 #define HDA_QUAD 4
+#define HDA_MAX 8
 
 static const struct snd_pcm_hardware azx_pcm_hw = {
 	.info =			(SNDRV_PCM_INFO_MMAP |
@@ -494,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 							stream->lpib);
 			snd_hdac_ext_stream_set_lpib(stream, stream->lpib);
 		}
+		/* fall through */
 
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -569,7 +571,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream,
 	stream_tag = hdac_stream(link_dev)->stream_tag;
 
 	/* set the stream tag in the codec dai dma params  */
-	snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0);
+	else
+		snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0);
 
 	p_params.s_fmt = snd_pcm_format_width(params_format(params));
 	p_params.ch = params_channels(params);
@@ -995,21 +1000,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = {
 	},
 },
 {
-	.name = "HD-Codec Pin",
+	.name = "Analog CPU DAI",
 	.ops = &skl_link_dai_ops,
 	.playback = {
-		.stream_name = "HD-Codec Tx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.stream_name = "Analog CPU Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
 	},
 	.capture = {
-		.stream_name = "HD-Codec Rx",
-		.channels_min = HDA_STEREO,
-		.channels_max = HDA_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.stream_name = "Analog CPU Capture",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+},
+{
+	.name = "Alt Analog CPU DAI",
+	.ops = &skl_link_dai_ops,
+	.playback = {
+		.stream_name = "Alt Analog CPU Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.stream_name = "Alt Analog CPU Capture",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+},
+{
+	.name = "Digital CPU DAI",
+	.ops = &skl_link_dai_ops,
+	.playback = {
+		.stream_name = "Digital CPU Playback",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.stream_name = "Digital CPU Capture",
+		.channels_min = HDA_MONO,
+		.channels_max = HDA_MAX,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+			SNDRV_PCM_FMTBIT_S32_LE,
 	},
 },
 };
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 2620d77729c5..cf8848b779dc 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w,
 			bc = (struct skl_algo_data *)sb->dobj.private;
 
 			if (bc->set_params == SKL_PARAM_BIND) {
-				params = kzalloc(bc->max, GFP_KERNEL);
+				params = kmemdup(bc->params, bc->max, GFP_KERNEL);
 				if (!params)
 					return -ENOMEM;
 
-				memcpy(params, bc->params, bc->max);
 				skl_fill_sink_instance_id(ctx, params, bc->max,
 								mconfig);
 
@@ -2461,6 +2460,7 @@ static int skl_tplg_get_token(struct device *dev,
 
 	case SKL_TKN_U8_CORE_ID:
 		mconfig->core_id = tkn_elem->value;
+		break;
 
 	case SKL_TKN_U8_MOD_TYPE:
 		mconfig->m_type = tkn_elem->value;
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 1d17be0f78a0..29225623b4b4 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -33,9 +33,11 @@
 #include <sound/hda_register.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_i915.h>
+#include <sound/hda_codec.h>
 #include "skl.h"
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
+#include "../../../soc/codecs/hdac_hda.h"
 
 /*
  * initialize the PCI registers
@@ -472,6 +474,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = {
 						{.name = "ssp5_sclkfs"},
 };
 
+static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl,
+					struct snd_soc_acpi_mach *machines)
+{
+	struct hdac_bus *bus = skl_to_bus(skl);
+	struct snd_soc_acpi_mach *mach;
+
+	/* check if we have any codecs detected on bus */
+	if (bus->codec_mask == 0)
+		return NULL;
+
+	/* point to common table */
+	mach = snd_soc_acpi_intel_hda_machines;
+
+	/* all entries in the machine table use the same firmware */
+	mach->fw_filename = machines->fw_filename;
+
+	return mach;
+}
+
 static int skl_find_machine(struct skl *skl, void *driver_data)
 {
 	struct hdac_bus *bus = skl_to_bus(skl);
@@ -479,9 +500,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 	struct skl_machine_pdata *pdata;
 
 	mach = snd_soc_acpi_find_machine(mach);
-	if (mach == NULL) {
-		dev_err(bus->dev, "No matching machine driver found\n");
-		return -ENODEV;
+	if (!mach) {
+		dev_dbg(bus->dev, "No matching I2S machine driver found\n");
+		mach = skl_find_hda_machine(skl, driver_data);
+		if (!mach) {
+			dev_err(bus->dev, "No matching machine driver found\n");
+			return -ENODEV;
+		}
 	}
 
 	skl->mach = mach;
@@ -498,8 +523,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data)
 
 static int skl_machine_device_register(struct skl *skl)
 {
-	struct hdac_bus *bus = skl_to_bus(skl);
 	struct snd_soc_acpi_mach *mach = skl->mach;
+	struct hdac_bus *bus = skl_to_bus(skl);
+	struct skl_machine_pdata *pdata;
 	struct platform_device *pdev;
 	int ret;
 
@@ -516,8 +542,12 @@ static int skl_machine_device_register(struct skl *skl)
 		return -EIO;
 	}
 
-	if (mach->pdata)
+	if (mach->pdata) {
+		pdata = (struct skl_machine_pdata *)mach->pdata;
+		pdata->platform = dev_name(bus->dev);
+		pdata->codec_mask = bus->codec_mask;
 		dev_set_drvdata(&pdev->dev, mach->pdata);
+	}
 
 	skl->i2s_dev = pdev;
 
@@ -628,6 +658,24 @@ static void skl_clock_device_unregister(struct skl *skl)
 		platform_device_unregister(skl->clk_dev);
 }
 
+#define IDISP_INTEL_VENDOR_ID	0x80860000
+
+/*
+ * load the legacy codec driver
+ */
+static void load_codec_module(struct hda_codec *codec)
+{
+#ifdef MODULE
+	char modalias[MODULE_NAME_LEN];
+	const char *mod = NULL;
+
+	snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+	mod = modalias;
+	dev_dbg(&codec->core.dev, "loading %s codec module\n", mod);
+	request_module(mod);
+#endif
+}
+
 /*
  * Probe the given codec address
  */
@@ -637,7 +685,9 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res = -1;
 	struct skl *skl = bus_to_skl(bus);
+	struct hdac_hda_priv *hda_codec;
 	struct hdac_device *hdev;
+	int err;
 
 	mutex_lock(&bus->cmd_mutex);
 	snd_hdac_bus_send_cmd(bus, cmd);
@@ -645,13 +695,26 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 	mutex_unlock(&bus->cmd_mutex);
 	if (res == -1)
 		return -EIO;
-	dev_dbg(bus->dev, "codec #%d probed OK\n", addr);
+	dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
 
-	hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
-	if (!hdev)
+	hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
+				 GFP_KERNEL);
+	if (!hda_codec)
 		return -ENOMEM;
 
-	return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+	hda_codec->codec.bus = skl_to_hbus(skl);
+	hdev = &hda_codec->codec.core;
+
+	err = snd_hdac_ext_bus_device_init(bus, addr, hdev);
+	if (err < 0)
+		return err;
+
+	/* use legacy bus only for HDA codecs, idisp uses ext bus */
+	if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) {
+		hdev->type = HDA_DEV_LEGACY;
+		load_codec_module(&hda_codec->codec);
+	}
+	return 0;
 }
 
 /* Codec initialization */
@@ -786,9 +849,10 @@ static int skl_create(struct pci_dev *pci,
 		      const struct hdac_io_ops *io_ops,
 		      struct skl **rskl)
 {
+	struct hdac_ext_bus_ops *ext_ops = NULL;
 	struct skl *skl;
 	struct hdac_bus *bus;
-
+	struct hda_bus *hbus;
 	int err;
 
 	*rskl = NULL;
@@ -803,13 +867,23 @@ static int skl_create(struct pci_dev *pci,
 		return -ENOMEM;
 	}
 
+	hbus = skl_to_hbus(skl);
 	bus = skl_to_bus(skl);
-	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, NULL);
+
+#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+	ext_ops = snd_soc_hdac_hda_get_ops();
+#endif
+	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
 	bus->use_posbuf = 1;
 	skl->pci = pci;
 	INIT_WORK(&skl->probe_work, skl_probe_work);
 	bus->bdl_pos_adj = 0;
 
+	mutex_init(&hbus->prepare_mutex);
+	hbus->pci = pci;
+	hbus->mixer_assigned = -1;
+	hbus->modelname = "sklbus";
+
 	*rskl = skl;
 
 	return 0;
diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h
index 78aa8bdcb619..8d48cd7c56c8 100644
--- a/sound/soc/intel/skylake/skl.h
+++ b/sound/soc/intel/skylake/skl.h
@@ -23,6 +23,7 @@
 
 #include <sound/hda_register.h>
 #include <sound/hdaudio_ext.h>
+#include <sound/hda_codec.h>
 #include <sound/soc.h>
 #include "skl-nhlt.h"
 #include "skl-ssp-clk.h"
@@ -71,7 +72,7 @@ struct skl_fw_config {
 };
 
 struct skl {
-	struct hdac_bus hbus;
+	struct hda_bus hbus;
 	struct pci_dev *pci;
 
 	unsigned int init_done:1; /* delayed init status */
@@ -105,8 +106,11 @@ struct skl {
 	struct snd_soc_acpi_mach *mach;
 };
 
-#define skl_to_bus(s)  (&(s)->hbus)
-#define bus_to_skl(bus) container_of(bus, struct skl, hbus)
+#define skl_to_bus(s)  (&(s)->hbus.core)
+#define bus_to_skl(bus) container_of(bus, struct skl, hbus.core)
+
+#define skl_to_hbus(s) (&(s)->hbus)
+#define hbus_to_skl(hbus) container_of((hbus), struct skl, (hbus))
 
 /* to pass dai dma data */
 struct skl_dma_params {
@@ -117,6 +121,8 @@ struct skl_dma_params {
 struct skl_machine_pdata {
 	u32 dmic_num;
 	bool use_tplg_pcm; /* use dais and dai links from topology */
+	const char *platform;
+	u32 codec_mask;
 };
 
 struct skl_dsp_ops {
diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
index 666282b865a8..97f9f38ce6b3 100644
--- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c
+++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c
@@ -299,6 +299,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
 		devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private),
 			     GFP_KERNEL);
 	struct device *dev = &pdev->dev;
+	struct snd_soc_dai_link *dai_link;
 
 	if (!priv)
 		return -ENOMEM;
@@ -309,10 +310,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_cs42448_dai_links[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt2701_cs42448_dai_links[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	card->dev = dev;
@@ -324,10 +325,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_cs42448_dai_links[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt2701_cs42448_dai_links[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 
 	codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node,
diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
index 89f34efd9747..6bc1d3d58e64 100644
--- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c
+++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c
@@ -97,6 +97,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt2701_wm8960_card;
 	struct device_node *platform_node, *codec_node;
+	struct snd_soc_dai_link *dai_link;
 	int ret, i;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -105,10 +106,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_wm8960_dai_links[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt2701_wm8960_dai_links[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	card->dev = &pdev->dev;
@@ -120,10 +121,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt2701_wm8960_dai_links[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt2701_wm8960_dai_links[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 
 	ret = snd_soc_of_parse_audio_routing(card, "audio-routing");
@@ -150,7 +151,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = {
 static struct platform_driver mt2701_wm8960_machine = {
 	.driver = {
 		.name = "mt2701-wm8960",
-		.owner = THIS_MODULE,
 #ifdef CONFIG_OF
 		.of_match_table = mt2701_wm8960_machine_dt_match,
 #endif
diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
index b1558c57b9ca..cc41eb531653 100644
--- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c
+++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c
@@ -158,6 +158,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt6797_mt6351_card;
 	struct device_node *platform_node, *codec_node;
+	struct snd_soc_dai_link *dai_link;
 	int ret, i;
 
 	card->dev = &pdev->dev;
@@ -168,10 +169,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt6797_mt6351_dai_links[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt6797_mt6351_dai_links[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -181,10 +182,10 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt6797_mt6351_dai_links[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt6797_mt6351_dai_links[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
@@ -205,7 +206,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = {
 static struct platform_driver mt6797_mt6351_driver = {
 	.driver = {
 		.name = "mt6797-mt6351",
-		.owner = THIS_MODULE,
 #ifdef CONFIG_OF
 		.of_match_table = mt6797_mt6351_dt_match,
 #endif
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 902d111016d6..4d6596d5cb07 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -137,6 +137,7 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_max98090_card;
 	struct device_node *codec_node, *platform_node;
+	struct snd_soc_dai_link *dai_link;
 	int ret, i;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -145,10 +146,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Property 'platform' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_max98090_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_max98090_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	codec_node = of_parse_phandle(pdev->dev.of_node,
@@ -158,10 +159,10 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
 			"Property 'audio-codec' missing or invalid\n");
 		return -EINVAL;
 	}
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_max98090_dais[i].codec_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->codec_name)
 			continue;
-		mt8173_max98090_dais[i].codec_of_node = codec_node;
+		dai_link->codec_of_node = codec_node;
 	}
 	card->dev = &pdev->dev;
 
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 582174d98c6c..da5b58ce791b 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -44,11 +44,10 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream,
 					  struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* pll from mclk 12.288M */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
 					  params_rate(params) * 512);
@@ -179,6 +178,7 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_rt5650_rt5514_card;
 	struct device_node *platform_node;
+	struct snd_soc_dai_link *dai_link;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -188,10 +188,10 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_rt5650_rt5514_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	mt8173_rt5650_rt5514_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index b3670c8a5b8d..d83cd039b413 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -48,11 +48,10 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream,
 					  struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* pll from mclk 12.288M */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
 					  params_rate(params) * 512);
@@ -225,6 +224,7 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
 	struct device_node *platform_node;
+	struct snd_soc_dai_link *dai_link;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -234,10 +234,10 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_rt5650_rt5676_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	mt8173_rt5650_rt5676_codecs[0].of_node =
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index 7a89b4aad182..7edf250c8fb1 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -59,6 +59,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	unsigned int mclk_clock;
+	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
 	switch (mt8173_rt5650_priv.pll_from) {
@@ -76,9 +77,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* pll from mclk */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock,
 					  params_rate(params) * 512);
@@ -240,6 +239,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 	struct device_node *platform_node;
 	struct device_node *np;
 	const char *codec_capture_dai;
+	struct snd_soc_dai_link *dai_link;
 	int i, ret;
 
 	platform_node = of_parse_phandle(pdev->dev.of_node,
@@ -249,10 +249,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	for (i = 0; i < card->num_links; i++) {
-		if (mt8173_rt5650_dais[i].platform_name)
+	for_each_card_prelinks(card, i, dai_link) {
+		if (dai_link->platform_name)
 			continue;
-		mt8173_rt5650_dais[i].platform_of_node = platform_node;
+		dai_link->platform_of_node = platform_node;
 	}
 
 	mt8173_rt5650_codecs[0].of_node =
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 8af8bc358a90..8b8426ed2363 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms"
 config SND_MESON_AXG_FIFO
 	tristate
 	select REGMAP_MMIO
+	imply COMMON_CLK_AXG_AUDIO
+	imply RESET_MESON_AUDIO_ARB
 
 config SND_MESON_AXG_FRDDR
 	tristate "Amlogic AXG Playback FIFO support"
@@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR
 config SND_MESON_AXG_TDM_FORMATTER
 	tristate
 	select REGMAP_MMIO
+	imply COMMON_CLK_AXG_AUDIO
 
 config SND_MESON_AXG_TDM_INTERFACE
 	tristate
@@ -51,6 +54,7 @@ config SND_MESON_AXG_SOUND_CARD
 	imply SND_MESON_AXG_TDMIN
 	imply SND_MESON_AXG_TDMOUT
 	imply SND_MESON_AXG_SPDIFOUT
+	imply SND_MESON_AXG_PDM
 	help
 	  Select Y or M to add support for the AXG SoC sound card
 
@@ -58,8 +62,17 @@ config SND_MESON_AXG_SPDIFOUT
 	tristate "Amlogic AXG SPDIF Output Support"
 	select SND_PCM_IEC958
 	imply SND_SOC_SPDIF
+	imply COMMON_CLK_AXG_AUDIO
 	help
 	  Select Y or M to add support for SPDIF output serializer embedded
 	  in the Amlogic AXG SoC family
 
+config SND_MESON_AXG_PDM
+	tristate "Amlogic AXG PDM Input Support"
+	imply SND_SOC_DMIC
+	imply COMMON_CLK_AXG_AUDIO
+	help
+	  Select Y or M to add support for PDM input embedded
+	  in the Amlogic AXG SoC family
+
 endmenu
diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile
index c5e003b093db..4cd25104029d 100644
--- a/sound/soc/meson/Makefile
+++ b/sound/soc/meson/Makefile
@@ -9,6 +9,7 @@ snd-soc-meson-axg-tdmin-objs := axg-tdmin.o
 snd-soc-meson-axg-tdmout-objs := axg-tdmout.o
 snd-soc-meson-axg-sound-card-objs := axg-card.o
 snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
+snd-soc-meson-axg-pdm-objs := axg-pdm.o
 
 obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
 obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
@@ -19,3 +20,4 @@ obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o
 obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o
 obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o
 obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
+obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 2914ba0d965b..aa54d2c612c9 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -97,14 +97,14 @@ static void axg_card_clean_references(struct axg_card *priv)
 {
 	struct snd_soc_card *card = &priv->card;
 	struct snd_soc_dai_link *link;
+	struct snd_soc_dai_link_component *codec;
 	int i, j;
 
 	if (card->dai_link) {
-		for (i = 0; i < card->num_links; i++) {
-			link = &card->dai_link[i];
+		for_each_card_prelinks(card, i, link) {
 			of_node_put(link->cpu_of_node);
-			for (j = 0; j < link->num_codecs; j++)
-				of_node_put(link->codecs[j].of_node);
+			for_each_link_codecs(link, j, codec)
+				of_node_put(codec->of_node);
 		}
 	}
 
@@ -167,8 +167,7 @@ static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
 	if (be->mclk_fs) {
 		mclk = params_rate(params) * be->mclk_fs;
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
 						     SND_SOC_CLOCK_IN);
 			if (ret && ret != -ENOTSUPP)
@@ -196,8 +195,7 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_dai *codec_dai;
 	int ret, i;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_tdm_slot(codec_dai,
 					       be->codec_masks[i].tx,
 					       be->codec_masks[i].rx,
@@ -478,7 +476,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card,
 
 	ret = axg_card_set_link_name(card, link, "be");
 	if (ret)
-		dev_err(card->dev, "error setting %s link name\n", np->name);
+		dev_err(card->dev, "error setting %pOFn link name\n", np);
 
 	return ret;
 }
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 30262550e37b..0e4f65e654c4 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss)
 
 	ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0,
 			  dev_name(dev), ss);
+	if (ret)
+		return ret;
 
 	/* Enable pclk to access registers and clock the fifo ip */
 	ret = clk_prepare_enable(fifo->pclk);
diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c
new file mode 100644
index 000000000000..9d5684493ffc
--- /dev/null
+++ b/sound/soc/meson/axg-pdm.c
@@ -0,0 +1,654 @@
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2018 BayLibre, SAS.
+// Author: Jerome Brunet <jbrunet@baylibre.com>
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm_params.h>
+
+#define PDM_CTRL			0x00
+#define  PDM_CTRL_EN			BIT(31)
+#define  PDM_CTRL_OUT_MODE		BIT(29)
+#define  PDM_CTRL_BYPASS_MODE		BIT(28)
+#define  PDM_CTRL_RST_FIFO		BIT(16)
+#define  PDM_CTRL_CHAN_RSTN_MASK	GENMASK(15, 8)
+#define  PDM_CTRL_CHAN_RSTN(x)		((x) << 8)
+#define  PDM_CTRL_CHAN_EN_MASK		GENMASK(7, 0)
+#define  PDM_CTRL_CHAN_EN(x)		((x) << 0)
+#define PDM_HCIC_CTRL1			0x04
+#define  PDM_FILTER_EN			BIT(31)
+#define  PDM_HCIC_CTRL1_GAIN_SFT_MASK	GENMASK(29, 24)
+#define  PDM_HCIC_CTRL1_GAIN_SFT(x)	((x) << 24)
+#define  PDM_HCIC_CTRL1_GAIN_MULT_MASK	GENMASK(23, 16)
+#define  PDM_HCIC_CTRL1_GAIN_MULT(x)	((x) << 16)
+#define  PDM_HCIC_CTRL1_DSR_MASK	GENMASK(8, 4)
+#define  PDM_HCIC_CTRL1_DSR(x)		((x) << 4)
+#define  PDM_HCIC_CTRL1_STAGE_NUM_MASK	GENMASK(3, 0)
+#define  PDM_HCIC_CTRL1_STAGE_NUM(x)	((x) << 0)
+#define PDM_HCIC_CTRL2			0x08
+#define PDM_F1_CTRL			0x0c
+#define  PDM_LPF_ROUND_MODE_MASK	GENMASK(17, 16)
+#define  PDM_LPF_ROUND_MODE(x)		((x) << 16)
+#define  PDM_LPF_DSR_MASK		GENMASK(15, 12)
+#define  PDM_LPF_DSR(x)			((x) << 12)
+#define  PDM_LPF_STAGE_NUM_MASK		GENMASK(8, 0)
+#define  PDM_LPF_STAGE_NUM(x)		((x) << 0)
+#define  PDM_LPF_MAX_STAGE		336
+#define  PDM_LPF_NUM			3
+#define PDM_F2_CTRL			0x10
+#define PDM_F3_CTRL			0x14
+#define PDM_HPF_CTRL			0x18
+#define  PDM_HPF_SFT_STEPS_MASK		GENMASK(20, 16)
+#define  PDM_HPF_SFT_STEPS(x)		((x) << 16)
+#define  PDM_HPF_OUT_FACTOR_MASK	GENMASK(15, 0)
+#define  PDM_HPF_OUT_FACTOR(x)		((x) << 0)
+#define PDM_CHAN_CTRL			0x1c
+#define  PDM_CHAN_CTRL_POINTER_WIDTH	8
+#define  PDM_CHAN_CTRL_POINTER_MAX	((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1)
+#define  PDM_CHAN_CTRL_NUM		4
+#define PDM_CHAN_CTRL1			0x20
+#define PDM_COEFF_ADDR			0x24
+#define PDM_COEFF_DATA			0x28
+#define PDM_CLKG_CTRL			0x2c
+#define PDM_STS				0x30
+
+struct axg_pdm_lpf {
+	unsigned int ds;
+	unsigned int round_mode;
+	const unsigned int *tap;
+	unsigned int tap_num;
+};
+
+struct axg_pdm_hcic {
+	unsigned int shift;
+	unsigned int mult;
+	unsigned int steps;
+	unsigned int ds;
+};
+
+struct axg_pdm_hpf {
+	unsigned int out_factor;
+	unsigned int steps;
+};
+
+struct axg_pdm_filters {
+	struct axg_pdm_hcic hcic;
+	struct axg_pdm_hpf hpf;
+	struct axg_pdm_lpf lpf[PDM_LPF_NUM];
+};
+
+struct axg_pdm_cfg {
+	const struct axg_pdm_filters *filters;
+	unsigned int sys_rate;
+};
+
+struct axg_pdm {
+	const struct axg_pdm_cfg *cfg;
+	struct regmap *map;
+	struct clk *dclk;
+	struct clk *sysclk;
+	struct clk *pclk;
+};
+
+static void axg_pdm_enable(struct regmap *map)
+{
+	/* Reset AFIFO */
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO);
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0);
+
+	/* Enable PDM */
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN);
+}
+
+static void axg_pdm_disable(struct regmap *map)
+{
+	regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0);
+}
+
+static void axg_pdm_filters_enable(struct regmap *map, bool enable)
+{
+	unsigned int val = enable ? PDM_FILTER_EN : 0;
+
+	regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val);
+	regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val);
+}
+
+static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		axg_pdm_enable(priv->map);
+		return 0;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		axg_pdm_disable(priv->map);
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static unsigned int axg_pdm_get_os(struct axg_pdm *priv)
+{
+	const struct axg_pdm_filters *filters = priv->cfg->filters;
+	unsigned int os = filters->hcic.ds;
+	int i;
+
+	/*
+	 * The global oversampling factor is defined by the down sampling
+	 * factor applied by each filter (HCIC and LPFs)
+	 */
+
+	for (i = 0; i < PDM_LPF_NUM; i++)
+		os *= filters->lpf[i].ds;
+
+	return os;
+}
+
+static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os,
+			      unsigned int rate)
+{
+	unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX;
+
+	/*
+	 * Set the default system clock rate unless it is too fast for
+	 * for the requested sample rate. In this case, the sample pointer
+	 * counter could overflow so set a lower system clock rate
+	 */
+	if (sys_rate < priv->cfg->sys_rate)
+		return clk_set_rate(priv->sysclk, sys_rate);
+
+	return clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+}
+
+static int axg_pdm_set_sample_pointer(struct axg_pdm *priv)
+{
+	unsigned int spmax, sp, val;
+	int i;
+
+	/* Max sample counter value per half period of dclk */
+	spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk),
+				 clk_get_rate(priv->dclk) * 2);
+
+	/* Check if sysclk is not too fast - should not happen */
+	if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX))
+		return -EINVAL;
+
+	/* Capture the data when we are at 75% of the half period */
+	sp = spmax * 3 / 4;
+
+	for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++)
+		val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i);
+
+	regmap_write(priv->map, PDM_CHAN_CTRL, val);
+	regmap_write(priv->map, PDM_CHAN_CTRL1, val);
+
+	return 0;
+}
+
+static void axg_pdm_set_channel_mask(struct axg_pdm *priv,
+				     unsigned int channels)
+{
+	unsigned int mask = GENMASK(channels - 1, 0);
+
+	/* Put all channel in reset */
+	regmap_update_bits(priv->map, PDM_CTRL,
+			   PDM_CTRL_CHAN_RSTN_MASK, 0);
+
+	/* Take the necessary channels out of reset and enable them */
+	regmap_update_bits(priv->map, PDM_CTRL,
+			   PDM_CTRL_CHAN_RSTN_MASK |
+			   PDM_CTRL_CHAN_EN_MASK,
+			   PDM_CTRL_CHAN_RSTN(mask) |
+			   PDM_CTRL_CHAN_EN(mask));
+}
+
+static int axg_pdm_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+	unsigned int os = axg_pdm_get_os(priv);
+	unsigned int rate = params_rate(params);
+	unsigned int val;
+	int ret;
+
+	switch (params_width(params)) {
+	case 24:
+		val = PDM_CTRL_OUT_MODE;
+		break;
+	case 32:
+		val = 0;
+		break;
+	default:
+		dev_err(dai->dev, "unsupported sample width\n");
+		return -EINVAL;
+	}
+
+	regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val);
+
+	ret = axg_pdm_set_sysclk(priv, os, rate);
+	if (ret) {
+		dev_err(dai->dev, "failed to set system clock\n");
+		return ret;
+	}
+
+	ret = clk_set_rate(priv->dclk, rate * os);
+	if (ret) {
+		dev_err(dai->dev, "failed to set dclk\n");
+		return ret;
+	}
+
+	ret = axg_pdm_set_sample_pointer(priv);
+	if (ret) {
+		dev_err(dai->dev, "invalid clock setting\n");
+		return ret;
+	}
+
+	axg_pdm_set_channel_mask(priv, params_channels(params));
+
+	return 0;
+}
+
+static int axg_pdm_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = clk_prepare_enable(priv->dclk);
+	if (ret) {
+		dev_err(dai->dev, "enabling dclk failed\n");
+		return ret;
+	}
+
+	/* Enable the filters */
+	axg_pdm_filters_enable(priv->map, true);
+
+	return ret;
+}
+
+static void axg_pdm_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+	axg_pdm_filters_enable(priv->map, false);
+	clk_disable_unprepare(priv->dclk);
+}
+
+static const struct snd_soc_dai_ops axg_pdm_dai_ops = {
+	.trigger	= axg_pdm_trigger,
+	.hw_params	= axg_pdm_hw_params,
+	.startup	= axg_pdm_startup,
+	.shutdown	= axg_pdm_shutdown,
+};
+
+static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv)
+{
+	const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic;
+	unsigned int val;
+
+	val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps);
+	val |= PDM_HCIC_CTRL1_DSR(hcic->ds);
+	val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult);
+	val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift);
+
+	regmap_update_bits(priv->map, PDM_HCIC_CTRL1,
+			   PDM_HCIC_CTRL1_STAGE_NUM_MASK |
+			   PDM_HCIC_CTRL1_DSR_MASK |
+			   PDM_HCIC_CTRL1_GAIN_MULT_MASK |
+			   PDM_HCIC_CTRL1_GAIN_SFT_MASK,
+			   val);
+}
+
+static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index)
+{
+	const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index];
+	unsigned int offset = index * regmap_get_reg_stride(priv->map)
+		+ PDM_F1_CTRL;
+	unsigned int val;
+
+	val = PDM_LPF_STAGE_NUM(lpf->tap_num);
+	val |= PDM_LPF_DSR(lpf->ds);
+	val |= PDM_LPF_ROUND_MODE(lpf->round_mode);
+
+	regmap_update_bits(priv->map, offset,
+			   PDM_LPF_STAGE_NUM_MASK |
+			   PDM_LPF_DSR_MASK |
+			   PDM_LPF_ROUND_MODE_MASK,
+			   val);
+}
+
+static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv)
+{
+	const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf;
+	unsigned int val;
+
+	val = PDM_HPF_OUT_FACTOR(hpf->out_factor);
+	val |= PDM_HPF_SFT_STEPS(hpf->steps);
+
+	regmap_update_bits(priv->map, PDM_HPF_CTRL,
+			   PDM_HPF_OUT_FACTOR_MASK |
+			   PDM_HPF_SFT_STEPS_MASK,
+			   val);
+}
+
+static int axg_pdm_set_lpf_filters(struct axg_pdm *priv)
+{
+	const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf;
+	unsigned int count = 0;
+	int i, j;
+
+	for (i = 0; i < PDM_LPF_NUM; i++)
+		count += lpf[i].tap_num;
+
+	/* Make sure the coeffs fit in the memory */
+	if (count >= PDM_LPF_MAX_STAGE)
+		return -EINVAL;
+
+	/* Set the initial APB bus register address */
+	regmap_write(priv->map, PDM_COEFF_ADDR, 0);
+
+	/* Set the tap filter values of all 3 filters */
+	for (i = 0; i < PDM_LPF_NUM; i++) {
+		axg_pdm_set_lpf_ctrl(priv, i);
+
+		for (j = 0; j < lpf[i].tap_num; j++)
+			regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]);
+	}
+
+	return 0;
+}
+
+static int axg_pdm_dai_probe(struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = clk_prepare_enable(priv->pclk);
+	if (ret) {
+		dev_err(dai->dev, "enabling pclk failed\n");
+		return ret;
+	}
+
+	/*
+	 * sysclk must be set and enabled as well to access the pdm registers
+	 * Accessing the register w/o it will give a bus error.
+	 */
+	ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate);
+	if (ret) {
+		dev_err(dai->dev, "setting sysclk failed\n");
+		goto err_pclk;
+	}
+
+	ret = clk_prepare_enable(priv->sysclk);
+	if (ret) {
+		dev_err(dai->dev, "enabling sysclk failed\n");
+		goto err_pclk;
+	}
+
+	/* Make sure the device is initially disabled */
+	axg_pdm_disable(priv->map);
+
+	/* Make sure filter bypass is disabled */
+	regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0);
+
+	/* Load filter settings */
+	axg_pdm_set_hcic_ctrl(priv);
+	axg_pdm_set_hpf_ctrl(priv);
+
+	ret = axg_pdm_set_lpf_filters(priv);
+	if (ret) {
+		dev_err(dai->dev, "invalid filter configuration\n");
+		goto err_sysclk;
+	}
+
+	return 0;
+
+err_sysclk:
+	clk_disable_unprepare(priv->sysclk);
+err_pclk:
+	clk_disable_unprepare(priv->pclk);
+	return ret;
+}
+
+static int axg_pdm_dai_remove(struct snd_soc_dai *dai)
+{
+	struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(priv->sysclk);
+	clk_disable_unprepare(priv->pclk);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver axg_pdm_dai_drv = {
+	.name = "PDM",
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 8,
+		.rates		= SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min	= 5512,
+		.rate_max	= 48000,
+		.formats	= (SNDRV_PCM_FMTBIT_S24_LE |
+				   SNDRV_PCM_FMTBIT_S32_LE),
+	},
+	.ops		= &axg_pdm_dai_ops,
+	.probe		= axg_pdm_dai_probe,
+	.remove		= axg_pdm_dai_remove,
+};
+
+static const struct snd_soc_component_driver axg_pdm_component_drv = {};
+
+static const struct regmap_config axg_pdm_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= PDM_STS,
+};
+
+static const unsigned int lpf1_default_tap[] = {
+	0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45,
+	0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21,
+	0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212,
+	0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa,
+	0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef,
+	0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077,
+	0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690,
+	0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3,
+	0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686,
+	0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812,
+	0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60,
+	0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000,
+	0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e,
+	0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c,
+	0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece,
+	0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645,
+	0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9,
+	0xffffb2, 0x000014,
+};
+
+static const unsigned int lpf2_default_tap[] = {
+	0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818,
+	0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41,
+	0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0,
+	0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000,
+	0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93,
+	0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12,
+	0x0002c1, 0xfff004, 0x00050a,
+};
+
+static const unsigned int lpf3_default_tap[] = {
+	0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000,
+	0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7,
+	0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000,
+	0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0,
+	0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000,
+	0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7,
+	0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000,
+	0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45,
+	0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000,
+	0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c,
+	0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000,
+	0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216,
+	0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000,
+	0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d,
+	0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000,
+	0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368,
+	0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000,
+	0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb,
+	0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000,
+	0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c,
+	0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000,
+	0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46,
+	0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000,
+	0x000081, 0x000000,
+};
+
+/*
+ * These values are sane defaults for the axg platform:
+ * - OS = 64
+ * - Latency = 38700 (?)
+ *
+ * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible.
+ *       the configuration may depend on the dmic used by the platform, the
+ *       expected tradeoff between latency and quality, etc ... If/When other
+ *       settings are required, we should add a fw interface to this driver to
+ *       load new filter settings.
+ */
+static const struct axg_pdm_filters axg_default_filters = {
+	.hcic = {
+		.shift = 0x15,
+		.mult = 0x80,
+		.steps = 7,
+		.ds = 8,
+	},
+	.hpf = {
+		.out_factor = 0x8000,
+		.steps = 13,
+	},
+	.lpf = {
+		[0] = {
+			.ds = 2,
+			.round_mode = 1,
+			.tap = lpf1_default_tap,
+			.tap_num = ARRAY_SIZE(lpf1_default_tap),
+		},
+		[1] = {
+			.ds = 2,
+			.round_mode = 0,
+			.tap = lpf2_default_tap,
+			.tap_num = ARRAY_SIZE(lpf2_default_tap),
+		},
+		[2] = {
+			.ds = 2,
+			.round_mode = 1,
+			.tap = lpf3_default_tap,
+			.tap_num = ARRAY_SIZE(lpf3_default_tap)
+		},
+	},
+};
+
+static const struct axg_pdm_cfg axg_pdm_config = {
+	.filters = &axg_default_filters,
+	.sys_rate = 250000000,
+};
+
+static const struct of_device_id axg_pdm_of_match[] = {
+	{
+		.compatible = "amlogic,axg-pdm",
+		.data = &axg_pdm_config,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, axg_pdm_of_match);
+
+static int axg_pdm_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct axg_pdm *priv;
+	struct resource *res;
+	void __iomem *regs;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->cfg = of_device_get_match_data(dev);
+	if (!priv->cfg) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	priv->map = devm_regmap_init_mmio(dev, regs, &axg_pdm_regmap_cfg);
+	if (IS_ERR(priv->map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(priv->map));
+		return PTR_ERR(priv->map);
+	}
+
+	priv->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(priv->pclk)) {
+		ret = PTR_ERR(priv->pclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get pclk: %d\n", ret);
+		return ret;
+	}
+
+	priv->dclk = devm_clk_get(dev, "dclk");
+	if (IS_ERR(priv->dclk)) {
+		ret = PTR_ERR(priv->dclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get dclk: %d\n", ret);
+		return ret;
+	}
+
+	priv->sysclk = devm_clk_get(dev, "sysclk");
+	if (IS_ERR(priv->sysclk)) {
+		ret = PTR_ERR(priv->sysclk);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "failed to get dclk: %d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(dev, &axg_pdm_component_drv,
+					       &axg_pdm_dai_drv, 1);
+}
+
+static struct platform_driver axg_pdm_pdrv = {
+	.probe = axg_pdm_probe,
+	.driver = {
+		.name = "axg-pdm",
+		.of_match_table = axg_pdm_of_match,
+	},
+};
+module_platform_driver(axg_pdm_pdrv);
+
+MODULE_DESCRIPTION("Amlogic AXG PDM Input driver");
+MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 7b8baf46d968..585ce030b79b 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
 	struct axg_tdm_stream *rx = (struct axg_tdm_stream *)
 		dai->capture_dma_data;
 	unsigned int tx_slots, rx_slots;
+	unsigned int fmt = 0;
 
 	tx_slots = axg_tdm_slots_total(tx_mask);
 	rx_slots = axg_tdm_slots_total(rx_mask);
@@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask,
 		return -EINVAL;
 	}
 
-	/*
-	 * Amend the dai driver channel number and let dpcm channel merge do
-	 * its job
-	 */
-	if (tx) {
-		tx->mask = tx_mask;
-		dai->driver->playback.channels_max = tx_slots;
-	}
-
-	if (rx) {
-		rx->mask = rx_mask;
-		dai->driver->capture.channels_max = rx_slots;
-	}
-
 	iface->slots = slots;
 
 	switch (slot_width) {
 	case 0:
-		/* defaults width to 32 if not provided */
-		iface->slot_width = 32;
-		break;
-	case 8:
-	case 16:
-	case 24:
+		slot_width = 32;
+		/* Fall-through */
 	case 32:
-		iface->slot_width = slot_width;
+		fmt |= SNDRV_PCM_FMTBIT_S32_LE;
+		/* Fall-through */
+	case 24:
+		fmt |= SNDRV_PCM_FMTBIT_S24_LE;
+		fmt |= SNDRV_PCM_FMTBIT_S20_LE;
+		/* Fall-through */
+	case 16:
+		fmt |= SNDRV_PCM_FMTBIT_S16_LE;
+		/* Fall-through */
+	case 8:
+		fmt |= SNDRV_PCM_FMTBIT_S8;
 		break;
 	default:
 		dev_err(dai->dev, "unsupported slot width: %d\n", slot_width);
 		return -EINVAL;
 	}
 
+	iface->slot_width = slot_width;
+
+	/* Amend the dai driver and let dpcm merge do its job */
+	if (tx) {
+		tx->mask = tx_mask;
+		dai->driver->playback.channels_max = tx_slots;
+		dai->driver->playback.formats = fmt;
+	}
+
+	if (rx) {
+		rx->mask = rx_mask;
+		dai->driver->capture.channels_max = rx_slots;
+		dai->driver->capture.formats = fmt;
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 81b09d740ed9..6384bb6dacfd 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev)
 	if (ret)
 		goto out;
 
-	ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
+	ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component,
 					 &nuc900_ac97_dai, 1);
 	if (ret)
 		goto out;
@@ -373,8 +373,6 @@ out:
 
 static int nuc900_ac97_drvremove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
-
 	nuc900_ac97_data = NULL;
 	snd_soc_set_ac97_ops(NULL);
 
diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index 8a99a8837dc9..673a9eb153b2 100644
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 	default:
 		return -EINVAL;
 	}
-	ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
+	ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
 					 dai_drv, 1);
 	if (ret)
 		return ret;
@@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(dev, "snd_soc_register_card failed (%d)\n", ret);
-		snd_soc_unregister_component(ad->dssdev);
 		return ret;
 	}
 
@@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev)
 	struct hdmi_audio_data *ad = platform_get_drvdata(pdev);
 
 	snd_soc_unregister_card(ad->card);
-	snd_soc_unregister_component(ad->dssdev);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 776e148b0aa2..943b44de1464 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -19,14 +19,13 @@ config SND_MMP_SOC
 
 config SND_PXA2XX_AC97
 	tristate
-	select SND_AC97_CODEC
 
 config SND_PXA2XX_SOC_AC97
 	tristate
-	select AC97_BUS
+	select AC97_BUS_NEW
 	select SND_PXA2XX_LIB
 	select SND_PXA2XX_LIB_AC97
-	select SND_SOC_AC97_BUS
+	select SND_SOC_AC97_BUS_NEW
 
 config SND_PXA2XX_SOC_I2S
 	select SND_PXA2XX_LIB
@@ -80,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA
 	tristate "SoC AC97 Audio support for Tosa"
 	depends on SND_PXA2XX_SOC && MACH_TOSA
 	depends on MFD_TC6393XB
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -89,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA
 config SND_PXA2XX_SOC_E740
 	tristate "SoC AC97 Audio support for e740"
 	depends on SND_PXA2XX_SOC && MACH_E740
+	depends on !AC97_BUS
 	select SND_SOC_WM9705
 	select SND_PXA2XX_SOC_AC97
 	help
@@ -98,6 +99,7 @@ config SND_PXA2XX_SOC_E740
 config SND_PXA2XX_SOC_E750
 	tristate "SoC AC97 Audio support for e750"
 	depends on SND_PXA2XX_SOC && MACH_E750
+	depends on !AC97_BUS
 	select SND_SOC_WM9705
 	select SND_PXA2XX_SOC_AC97
 	help
@@ -107,6 +109,7 @@ config SND_PXA2XX_SOC_E750
 config SND_PXA2XX_SOC_E800
 	tristate "SoC AC97 Audio support for e800"
 	depends on SND_PXA2XX_SOC && MACH_E800
+	depends on !AC97_BUS
 	select SND_SOC_WM9712
 	select SND_PXA2XX_SOC_AC97
 	help
@@ -117,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270
 	tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
 	depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
 			MACH_CM_X300)
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -127,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X
 	bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
 	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
 			MACH_PALMT5 || MACH_PALMTE2)
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -156,6 +161,7 @@ config SND_SOC_TTC_DKB
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+	depends on !AC97_BUS
 	select SND_PXA2XX_SOC_AC97
 	select SND_PXA_SOC_SSP
 	select SND_SOC_WM9713
@@ -195,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN
 config SND_PXA2XX_SOC_MIOA701
         tristate "SoC Audio support for MIO A701"
         depends on SND_PXA2XX_SOC && MACH_MIOA701
+	depends on !AC97_BUS
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9713
         help
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 69033e1a84e6..adcf8ba9d287 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		pxa_ssp_disable(ssp);
 	}
 
+	if (priv->extclk)
+		clk_prepare_enable(priv->extclk);
+
 	dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
 	if (!dma)
 		return -ENOMEM;
@@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 		clk_disable_unprepare(ssp->clk);
 	}
 
+	if (priv->extclk)
+		clk_disable_unprepare(priv->extclk);
+
 	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
 	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
 }
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9f779657bc86..f8a3aa6c6d4e 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -17,6 +17,7 @@
 #include <linux/dmaengine.h>
 #include <linux/dma/pxa-dma.h>
 
+#include <sound/ac97/controller.h>
 #include <sound/core.h>
 #include <sound/ac97_codec.h>
 #include <sound/soc.h>
@@ -27,43 +28,35 @@
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
-static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv)
 {
 	pxa2xx_ac97_try_warm_reset();
 
 	pxa2xx_ac97_finish_reset();
 }
 
-static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv)
 {
 	pxa2xx_ac97_try_cold_reset();
 
 	pxa2xx_ac97_finish_reset();
 }
 
-static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97,
-					      unsigned short reg)
+static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot,
+				  unsigned short reg)
 {
-	int ret;
-
-	ret = pxa2xx_ac97_read(ac97->num, reg);
-	if (ret < 0)
-		return 0;
-	else
-		return (unsigned short)(ret & 0xffff);
+	return pxa2xx_ac97_read(slot, reg);
 }
 
-static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97,
-				     unsigned short reg, unsigned short val)
+static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot,
+				   unsigned short reg, unsigned short val)
 {
-	int ret;
-
-	ret = pxa2xx_ac97_write(ac97->num, reg, val);
+	return pxa2xx_ac97_write(slot, reg, val);
 }
 
-static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
-	.read	= pxa2xx_ac97_legacy_read,
-	.write	= pxa2xx_ac97_legacy_write,
+static struct ac97_controller_ops pxa2xx_ac97_ops = {
+	.read	= pxa2xx_ac97_read_actrl,
+	.write	= pxa2xx_ac97_write_actrl,
 	.warm_reset	= pxa2xx_ac97_warm_reset,
 	.reset	= pxa2xx_ac97_cold_reset,
 };
@@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids);
 static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
 	int ret;
+	struct ac97_controller *ctrl;
+	pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data;
+	void **codecs_pdata;
 
 	if (pdev->id != -1) {
 		dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
@@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
-	if (ret != 0)
-		return ret;
+	codecs_pdata = pdata ? pdata->codec_pdata : NULL;
+	ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev,
+					    AC97_SLOTS_AVAILABLE_ALL,
+					    codecs_pdata);
+	if (IS_ERR(ctrl))
+		return PTR_ERR(ctrl);
 
+	platform_set_drvdata(pdev, ctrl);
 	/* Punt most of the init to the SoC probe; we may need the machine
 	 * driver to do interesting things with the clocking to get us up
 	 * and running.
@@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 
 static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
 {
+	struct ac97_controller *ctrl = platform_get_drvdata(pdev);
+
 	snd_soc_unregister_component(&pdev->dev);
-	snd_soc_set_ac97_ops(NULL);
+	snd_ac97_controller_unregister(ctrl);
 	pxa2xx_ac97_hw_remove(pdev);
 	return 0;
 }
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 1543e85629f8..fb45f396ab4a 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -25,13 +25,12 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 static void apq8096_add_be_ops(struct snd_soc_card *card)
 {
-	struct snd_soc_dai_link *link = card->dai_link;
-	int i, num_links = card->num_links;
+	struct snd_soc_dai_link *link;
+	int i;
 
-	for (i = 0; i < num_links; i++) {
+	for_each_card_prelinks(card, i, link) {
 		if (link->no_pcm == 1)
 			link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
-		link++;
 	}
 }
 
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
index 932c3ebfd252..da242515e146 100644
--- a/sound/soc/qcom/qdsp6/q6adm.c
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -2,25 +2,24 @@
 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
 // Copyright (c) 2018, Linaro Limited
 
-#include <linux/slab.h>
-#include <linux/wait.h>
-#include <linux/kernel.h>
 #include <linux/device.h>
-#include <linux/module.h>
-#include <linux/sched.h>
 #include <linux/jiffies.h>
+#include <linux/kernel.h>
+#include <linux/kref.h>
+#include <linux/module.h>
 #include <linux/of.h>
 #include <linux/of_platform.h>
-#include <linux/kref.h>
-#include <linux/wait.h>
-#include <linux/soc/qcom/apr.h>
 #include <linux/platform_device.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/wait.h>
 #include <sound/asound.h>
 #include "q6adm.h"
 #include "q6afe.h"
 #include "q6core.h"
-#include "q6dsp-errno.h"
 #include "q6dsp-common.h"
+#include "q6dsp-errno.h"
 
 #define ADM_CMD_DEVICE_OPEN_V5		0x00010326
 #define ADM_CMDRSP_DEVICE_OPEN_V5	0x00010329
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 9db9a2944ef2..a16c71c03058 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -8,7 +8,6 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <sound/soc.h>
-#include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
 #include <asm/dma.h>
@@ -319,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream)
 	prtd->audio_client = q6asm_audio_client_alloc(dev,
 				(q6asm_cb)event_handler, prtd, stream_id,
 				LEGACY_PCM_MODE);
-	if (!prtd->audio_client) {
+	if (IS_ERR(prtd->audio_client)) {
 		pr_info("%s: Could not allocate memory\n", __func__);
+		ret = PTR_ERR(prtd->audio_client);
 		kfree(prtd);
-		return -ENOMEM;
+		return ret;
 	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
 		}
 	}
 
-	return ret;
+	return 0;
 }
 
 static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 2b2c7233bb5f..e1cfa846a1dc 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,7 +11,6 @@
 #include <linux/spinlock.h>
 #include <linux/kref.h>
 #include <linux/of.h>
-#include <linux/of_platform.h>
 #include <uapi/sound/asound.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c
index 06f03a5fe9bd..cdfc8ab6cfc0 100644
--- a/sound/soc/qcom/qdsp6/q6core.c
+++ b/sound/soc/qcom/qdsp6/q6core.c
@@ -10,7 +10,6 @@
 #include <linux/of.h>
 #include <linux/of_platform.h>
 #include <linux/jiffies.h>
-#include <linux/wait.h>
 #include <linux/soc/qcom/apr.h>
 #include "q6core.h"
 #include "q6dsp-errno.h"
@@ -105,12 +104,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
 		bytes = sizeof(*fwk) + fwk->num_services *
 				sizeof(fwk->svc_api_info[0]);
 
-		core->fwk_version = kzalloc(bytes, GFP_ATOMIC);
+		core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC);
 		if (!core->fwk_version)
 			return -ENOMEM;
 
-		memcpy(core->fwk_version, data->payload, bytes);
-
 		core->fwk_version_supported = true;
 		core->resp_received = true;
 
@@ -124,12 +121,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data)
 
 		len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]);
 
-		core->svc_version = kzalloc(len, GFP_ATOMIC);
+		core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC);
 		if (!core->svc_version)
 			return -ENOMEM;
 
-		memcpy(core->svc_version, data->payload, len);
-
 		core->get_version_supported = true;
 		core->resp_received = true;
 
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 2a781d87ee65..9effbecc571f 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -195,15 +195,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 
 static void sdm845_add_be_ops(struct snd_soc_card *card)
 {
-	struct snd_soc_dai_link *link = card->dai_link;
-	int i, num_links = card->num_links;
+	struct snd_soc_dai_link *link;
+	int i;
 
-	for (i = 0; i < num_links; i++) {
+	for_each_card_prelinks(card, i, link) {
 		if (link->no_pcm == 1) {
 			link->ops = &sdm845_be_ops;
 			link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
 		}
-		link++;
 	}
 }
 
diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c
index 929b3fe289b0..a472d5eb2950 100644
--- a/sound/soc/rockchip/rk3288_hdmi_analog.c
+++ b/sound/soc/rockchip/rk3288_hdmi_analog.c
@@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = {
 	.probe = snd_rk_mc_probe,
 	.driver = {
 		.name = DRV_NAME,
-		.owner = THIS_MODULE,
 		.pm = &snd_soc_pm_ops,
 		.of_match_table = rockchip_sound_of_match,
 	},
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
index f77538319221..9e7b5fa4cf59 100644
--- a/sound/soc/rockchip/rockchip_pcm.c
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -21,7 +21,8 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_PAUSE |
-				  SNDRV_PCM_INFO_RESUME,
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_INTERLEAVED,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
 	.periods_min		= 1,
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index 43332c32d7e9..dc93941e01c3 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -491,6 +491,7 @@ static int tm2_probe(struct platform_device *pdev)
 	struct snd_soc_card *card = &tm2_card;
 	struct tm2_machine_priv *priv;
 	struct of_phandle_args args;
+	struct snd_soc_dai_link *dai_link;
 	int num_codecs, ret, i;
 
 	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
@@ -558,18 +559,18 @@ static int tm2_probe(struct platform_device *pdev)
 	}
 
 	/* Initialize WM5110 - I2S and HDMI - I2S1 DAI links */
-	for (i = 0; i < card->num_links; i++) {
+	for_each_card_prelinks(card, i, dai_link) {
 		unsigned int dai_index = 0; /* WM5110 */
 
-		card->dai_link[i].cpu_name = NULL;
-		card->dai_link[i].platform_name = NULL;
+		dai_link->cpu_name = NULL;
+		dai_link->platform_name = NULL;
 
 		if (num_codecs > 1 && i == card->num_links - 1)
 			dai_index = 1; /* HDMI */
 
-		card->dai_link[i].codec_of_node = codec_dai_node[dai_index];
-		card->dai_link[i].cpu_of_node = cpu_dai_node[dai_index];
-		card->dai_link[i].platform_of_node = cpu_dai_node[dai_index];
+		dai_link->codec_of_node = codec_dai_node[dai_index];
+		dai_link->cpu_of_node = cpu_dai_node[dai_index];
+		dai_link->platform_of_node = cpu_dai_node[dai_index];
 	}
 
 	if (num_codecs > 1) {
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c2b496398e6b..17622ceb98c0 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev)
 	if (ret != 0)
 		return ret;
 
-	return snd_soc_register_component(&pdev->dev, &sh4_hac_component,
+	return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component,
 					  sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
 }
 
 static int hac_soc_platform_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_set_ac97_ops(NULL);
 	return 0;
 }
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 051f96405346..28327dd2c6cb 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -582,7 +582,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
 	int i;
 
 	for_each_rsnd_clk(clk, adg, i)
-		dev_dbg(dev, "%s    : %p : %ld\n",
+		dev_dbg(dev, "%s    : %pa : %ld\n",
 			clk_name[i], clk, clk_get_rate(clk));
 
 	dev_dbg(dev, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n",
@@ -595,7 +595,7 @@ static void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct rsnd_adg *adg)
 	 * by BRGCKR::BRGCKR_31
 	 */
 	for_each_rsnd_clkout(clk, adg, i)
-		dev_dbg(dev, "clkout %d : %p : %ld\n", i,
+		dev_dbg(dev, "clkout %d : %pa : %ld\n", i,
 			clk, clk_get_rate(clk));
 }
 #else
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index d23c2bbff0cf..f930f51b686f 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -102,7 +102,9 @@
 #include "rsnd.h"
 
 #define RSND_RATES SNDRV_PCM_RATE_8000_192000
-#define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+#define RSND_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+		   SNDRV_PCM_FMTBIT_S16_LE |\
+		   SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct of_device_id rsnd_of_match[] = {
 	{ .compatible = "renesas,rcar_sound-gen1", .data = (void *)RSND_GEN1 },
@@ -280,6 +282,8 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 	struct device *dev = rsnd_priv_to_dev(priv);
 
 	switch (snd_pcm_format_width(runtime->format)) {
+	case 8:
+		return 16 << 16;
 	case 16:
 		return 8 << 16;
 	case 24:
@@ -331,7 +335,7 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
 		target = cmd ? cmd : ssiu;
 	}
 
-	/* Non target mod or 24bit data needs normal DALIGN */
+	/* Non target mod or non 16bit needs normal DALIGN */
 	if ((snd_pcm_format_width(runtime->format) != 16) ||
 	    (mod != target))
 		return 0x76543210;
@@ -367,7 +371,7 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
 	 * HW    24bit data is located as 0x******00
 	 *
 	 */
-	if (snd_pcm_format_width(runtime->format) == 16)
+	if (snd_pcm_format_width(runtime->format) != 24)
 		return 0;
 
 	for (i = 0; i < ARRAY_SIZE(playback_mods); i++) {
@@ -540,6 +544,14 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
 	return rdai->ssi_lane;
 }
 
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width)
+{
+	if (width > 0)
+		rdai->chan_width = width;
+
+	return rdai->chan_width;
+}
+
 struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id)
 {
 	if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
@@ -681,6 +693,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		rdai->frm_clk_inv = 0;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_DSP_B:
 		rdai->sys_delay = 1;
 		rdai->data_alignment = 0;
 		rdai->frm_clk_inv = 1;
@@ -690,6 +703,11 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		rdai->data_alignment = 1;
 		rdai->frm_clk_inv = 1;
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		rdai->sys_delay = 0;
+		rdai->data_alignment = 0;
+		rdai->frm_clk_inv = 1;
+		break;
 	}
 
 	/* set clock inversion */
@@ -720,6 +738,16 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
+	switch (slot_width) {
+	case 16:
+	case 24:
+	case 32:
+		break;
+	default:
+		/* use default */
+		slot_width = 32;
+	}
+
 	switch (slots) {
 	case 2:
 	case 6:
@@ -727,6 +755,7 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai,
 		/* TDM Extend Mode */
 		rsnd_rdai_channels_set(rdai, slots);
 		rsnd_rdai_ssi_lane_set(rdai, 1);
+		rsnd_rdai_width_set(rdai, slot_width);
 		break;
 	default:
 		dev_err(dev, "unsupported TDM slots (%d)\n", slots);
@@ -755,7 +784,7 @@ static unsigned int rsnd_soc_hw_rate_list[] = {
 	192000,
 };
 
-static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
+static int rsnd_soc_hw_rule(struct rsnd_dai *rdai,
 			    unsigned int *list, int list_num,
 			    struct snd_interval *baseline, struct snd_interval *iv)
 {
@@ -772,14 +801,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
 		if (!snd_interval_test(iv, list[i]))
 			continue;
 
-		rate = rsnd_ssi_clk_query(priv,
+		rate = rsnd_ssi_clk_query(rdai,
 					  baseline->min, list[i], NULL);
 		if (rate > 0) {
 			p.min = min(p.min, list[i]);
 			p.max = max(p.max, list[i]);
 		}
 
-		rate = rsnd_ssi_clk_query(priv,
+		rate = rsnd_ssi_clk_query(rdai,
 					  baseline->max, list[i], NULL);
 		if (rate > 0) {
 			p.min = min(p.min, list[i]);
@@ -790,17 +819,14 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv,
 	return snd_interval_refine(iv, &p);
 }
 
-static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
-				   struct snd_pcm_hw_rule *rule,
-				   int is_play)
+static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
+				 struct snd_pcm_hw_rule *rule)
 {
 	struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval ic;
-	struct snd_soc_dai *dai = rule->private;
-	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-	struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+	struct rsnd_dai_stream *io = rule->private;
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 
 	/*
 	 * possible sampling rate limitation is same as
@@ -811,34 +837,19 @@ static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params,
 	ic.min =
 	ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
 
-	return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list,
+	return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_rate_list,
 				ARRAY_SIZE(rsnd_soc_hw_rate_list),
 				&ic, ir);
 }
 
-static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params,
-				 struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_rate(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params,
-					  struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_rate(params, rule, 0);
-}
-
-static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
-				       struct snd_pcm_hw_rule *rule,
-				       int is_play)
+static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
+				     struct snd_pcm_hw_rule *rule)
 {
 	struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval ic;
-	struct snd_soc_dai *dai = rule->private;
-	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
-	struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture;
+	struct rsnd_dai_stream *io = rule->private;
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
 
 	/*
 	 * possible sampling rate limitation is same as
@@ -849,23 +860,11 @@ static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params,
 	ic.min =
 	ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params);
 
-	return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list,
+	return rsnd_soc_hw_rule(rdai, rsnd_soc_hw_channels_list,
 				ARRAY_SIZE(rsnd_soc_hw_channels_list),
 				ir, &ic);
 }
 
-static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params,
-					      struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_channels(params, rule, 1);
-}
-
-static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params,
-					     struct snd_pcm_hw_rule *rule)
-{
-	return __rsnd_soc_hw_rule_channels(params, rule, 0);
-}
-
 static const struct snd_pcm_hardware rsnd_pcm_hardware = {
 	.info =		SNDRV_PCM_INFO_INTERLEAVED	|
 			SNDRV_PCM_INFO_MMAP		|
@@ -882,12 +881,10 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai);
-	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
 	struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream);
 	struct snd_pcm_hw_constraint_list *constraint = &rdai->constraint;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int max_channels = rsnd_rdai_channels_get(rdai);
-	int ret;
 	int i;
 
 	rsnd_dai_stream_init(io, substream);
@@ -922,25 +919,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
 		int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				    is_play ? rsnd_soc_hw_rule_rate_playback :
-					      rsnd_soc_hw_rule_rate_capture,
-				    dai,
+				    rsnd_soc_hw_rule_rate,
+				    is_play ? &rdai->playback : &rdai->capture,
 				    SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-				    is_play ? rsnd_soc_hw_rule_channels_playback :
-					      rsnd_soc_hw_rule_channels_capture,
-				    dai,
+				    rsnd_soc_hw_rule_channels,
+				    is_play ? &rdai->playback : &rdai->capture,
 				    SNDRV_PCM_HW_PARAM_RATE, -1);
 	}
 
-	/*
-	 * call rsnd_dai_call without spinlock
-	 */
-	ret = rsnd_dai_call(nolock_start, io, priv);
-	if (ret < 0)
-		rsnd_dai_call(nolock_stop, io, priv);
-
-	return ret;
+	return 0;
 }
 
 static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
@@ -953,7 +941,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream,
 	/*
 	 * call rsnd_dai_call without spinlock
 	 */
-	rsnd_dai_call(nolock_stop, io, priv);
+	rsnd_dai_call(cleanup, io, priv);
 
 	rsnd_dai_stream_quit(io);
 }
@@ -1083,6 +1071,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv,
 	rdai->capture.rdai		= rdai;
 	rsnd_rdai_channels_set(rdai, 2); /* default 2ch */
 	rsnd_rdai_ssi_lane_set(rdai, 1); /* default 1lane */
+	rsnd_rdai_width_set(rdai, 32);   /* default 32bit width */
 
 	for (io_i = 0;; io_i++) {
 		playback = of_parse_phandle(dai_np, "playback", io_i);
@@ -1274,8 +1263,15 @@ int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io)
 int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io)
 {
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	struct rsnd_priv *priv = rsnd_io_to_priv(io);
+	struct device *dev = rsnd_priv_to_dev(priv);
+
+	if (!runtime) {
+		dev_warn(dev, "Can't update kctrl when idle\n");
+		return 0;
+	}
 
-	return !!runtime;
+	return 1;
 }
 
 struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg)
diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c
index 6a55aa753003..ad702377a6c3 100644
--- a/sound/soc/sh/rcar/ctu.c
+++ b/sound/soc/sh/rcar/ctu.c
@@ -258,7 +258,7 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod,
 		struct snd_pcm_hw_params *be_params;
 		int stream = substream->stream;
 
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		for_each_dpcm_be(fe, stream, dpcm) {
 			be_params = &dpcm->hw_params;
 			if (params_channels(fe_params) != params_channels(be_params))
 				ctu->channels = params_channels(be_params);
diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c
index d65ea7bc4dac..6d1947515dc8 100644
--- a/sound/soc/sh/rcar/dma.c
+++ b/sound/soc/sh/rcar/dma.c
@@ -106,9 +106,9 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod,
 	return 0;
 }
 
-static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
-				   struct rsnd_dai_stream *io,
-				   struct rsnd_priv *priv)
+static int rsnd_dmaen_cleanup(struct rsnd_mod *mod,
+			      struct rsnd_dai_stream *io,
+			      struct rsnd_priv *priv)
 {
 	struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
 	struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
@@ -116,7 +116,7 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
 	/*
 	 * DMAEngine release uses mutex lock.
 	 * Thus, it shouldn't be called under spinlock.
-	 * Let's call it under nolock_start
+	 * Let's call it under prepare
 	 */
 	if (dmaen->chan)
 		dma_release_channel(dmaen->chan);
@@ -126,23 +126,22 @@ static int rsnd_dmaen_nolock_stop(struct rsnd_mod *mod,
 	return 0;
 }
 
-static int rsnd_dmaen_nolock_start(struct rsnd_mod *mod,
-			    struct rsnd_dai_stream *io,
-			    struct rsnd_priv *priv)
+static int rsnd_dmaen_prepare(struct rsnd_mod *mod,
+			      struct rsnd_dai_stream *io,
+			      struct rsnd_priv *priv)
 {
 	struct rsnd_dma *dma = rsnd_mod_to_dma(mod);
 	struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
-	if (dmaen->chan) {
-		dev_err(dev, "it already has dma channel\n");
-		return -EIO;
-	}
+	/* maybe suspended */
+	if (dmaen->chan)
+		return 0;
 
 	/*
 	 * DMAEngine request uses mutex lock.
 	 * Thus, it shouldn't be called under spinlock.
-	 * Let's call it under nolock_start
+	 * Let's call it under prepare
 	 */
 	dmaen->chan = rsnd_dmaen_request_channel(io,
 						 dma->mod_from,
@@ -291,8 +290,8 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod,
 
 static struct rsnd_mod_ops rsnd_dmaen_ops = {
 	.name	= "audmac",
-	.nolock_start = rsnd_dmaen_nolock_start,
-	.nolock_stop  = rsnd_dmaen_nolock_stop,
+	.prepare = rsnd_dmaen_prepare,
+	.cleanup = rsnd_dmaen_cleanup,
 	.start	= rsnd_dmaen_start,
 	.stop	= rsnd_dmaen_stop,
 	.pointer= rsnd_dmaen_pointer,
@@ -302,16 +301,26 @@ static struct rsnd_mod_ops rsnd_dmaen_ops = {
  *		Audio DMAC peri peri
  */
 static const u8 gen2_id_table_ssiu[] = {
-	0x00, /* SSI00 */
-	0x04, /* SSI10 */
-	0x08, /* SSI20 */
-	0x0c, /* SSI3  */
-	0x0d, /* SSI4  */
-	0x0e, /* SSI5  */
-	0x0f, /* SSI6  */
-	0x10, /* SSI7  */
-	0x11, /* SSI8  */
-	0x12, /* SSI90 */
+	/* SSI00 ~ SSI07 */
+	0x00, 0x01, 0x02, 0x03, 0x39, 0x3a, 0x3b, 0x3c,
+	/* SSI10 ~ SSI17 */
+	0x04, 0x05, 0x06, 0x07, 0x3d, 0x3e, 0x3f, 0x40,
+	/* SSI20 ~ SSI27 */
+	0x08, 0x09, 0x0a, 0x0b, 0x41, 0x42, 0x43, 0x44,
+	/* SSI30 ~ SSI37 */
+	0x0c, 0x45, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b,
+	/* SSI40 ~ SSI47 */
+	0x0d, 0x4c, 0x4d, 0x4e, 0x4f, 0x50, 0x51, 0x52,
+	/* SSI5 */
+	0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI6 */
+	0x0f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI7 */
+	0x10, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI8 */
+	0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* SSI90 ~ SSI97 */
+	0x12, 0x13, 0x14, 0x15, 0x53, 0x54, 0x55, 0x56,
 };
 static const u8 gen2_id_table_scu[] = {
 	0x2d, /* SCU_SRCI0 */
@@ -337,18 +346,23 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io,
 	struct rsnd_mod *src = rsnd_io_to_mod_src(io);
 	struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
 	const u8 *entry = NULL;
-	int id = rsnd_mod_id(mod);
+	int id = 255;
 	int size = 0;
 
 	if (mod == ssi) {
+		int busif = rsnd_ssi_get_busif(io);
+
 		entry = gen2_id_table_ssiu;
 		size = ARRAY_SIZE(gen2_id_table_ssiu);
+		id = (rsnd_mod_id(mod) * 8) + busif;
 	} else if (mod == src) {
 		entry = gen2_id_table_scu;
 		size = ARRAY_SIZE(gen2_id_table_scu);
+		id = rsnd_mod_id(mod);
 	} else if (mod == dvc) {
 		entry = gen2_id_table_cmd;
 		size = ARRAY_SIZE(gen2_id_table_cmd);
+		id = rsnd_mod_id(mod);
 	}
 
 	if ((!entry) || (size <= id)) {
@@ -382,7 +396,7 @@ static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg)
 	struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv);
 	struct device *dev = rsnd_priv_to_dev(priv);
 
-	dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
+	dev_dbg(dev, "w 0x%px : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data);
 
 	iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg));
 }
@@ -491,11 +505,11 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = {
 #define RDMA_SSI_I_N(addr, i)	(addr ##_reg - 0x00300000 + (0x40 * i) + 0x8)
 #define RDMA_SSI_O_N(addr, i)	(addr ##_reg - 0x00300000 + (0x40 * i) + 0xc)
 
-#define RDMA_SSIU_I_N(addr, i)	(addr ##_reg - 0x00441000 + (0x1000 * i))
-#define RDMA_SSIU_O_N(addr, i)	(addr ##_reg - 0x00441000 + (0x1000 * i))
+#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j)
 
-#define RDMA_SSIU_I_P(addr, i)	(addr ##_reg - 0x00141000 + (0x1000 * i))
-#define RDMA_SSIU_O_P(addr, i)	(addr ##_reg - 0x00141000 + (0x1000 * i))
+#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400))
+#define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j)
 
 #define RDMA_SRC_I_N(addr, i)	(addr ##_reg - 0x00500000 + (0x400 * i))
 #define RDMA_SRC_O_N(addr, i)	(addr ##_reg - 0x004fc000 + (0x400 * i))
@@ -521,6 +535,7 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
 		      !!rsnd_io_to_mod_mix(io) ||
 		      !!rsnd_io_to_mod_ctu(io);
 	int id = rsnd_mod_id(mod);
+	int busif = rsnd_ssi_get_busif(io);
 	struct dma_addr {
 		dma_addr_t out_addr;
 		dma_addr_t in_addr;
@@ -537,25 +552,35 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io,
 		},
 		/* SSI */
 		/* Capture */
-		{{{ RDMA_SSI_O_N(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 } },
+		{{{ RDMA_SSI_O_N(ssi, id),		0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 } },
 		 /* Playback */
-		 {{ 0,				RDMA_SSI_I_N(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) } }
+		 {{ 0,			RDMA_SSI_I_N(ssi, id) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) } }
 		},
 		/* SSIU */
 		/* Capture */
-		{{{ RDMA_SSIU_O_N(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 },
-		  { RDMA_SSIU_O_P(ssi, id),	0 } },
+		{{{ RDMA_SSIU_O_N(ssi, id, busif),	0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 },
+		  { RDMA_SSIU_O_P(ssi, id, busif),	0 } },
 		 /* Playback */
-		 {{ 0,				RDMA_SSIU_I_N(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) },
-		  { 0,				RDMA_SSIU_I_P(ssi, id) } } },
+		 {{ 0,			RDMA_SSIU_I_N(ssi, id, busif) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) },
+		  { 0,			RDMA_SSIU_I_P(ssi, id, busif) } } },
 	};
 
+	/*
+	 * FIXME
+	 *
+	 * We can't support SSI9-4/5/6/7, because its address is
+	 * out of calculation rule
+	 */
+	if ((id == 9) && (busif >= 4))
+		dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+			id, busif);
+
 	/* it shouldn't happen */
 	if (use_cmd && !use_src)
 		dev_err(dev, "DVC is selected without SRC\n");
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 0230301fe078..1f7881cc16b2 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -219,12 +219,33 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv)
 		RSND_GEN_S_REG(HDMI1_SEL,	0x9e4),
 
 		/* FIXME: it needs SSI_MODE2/3 in the future */
-		RSND_GEN_M_REG(SSI_BUSIF_MODE,	0x0,	0x80),
-		RSND_GEN_M_REG(SSI_BUSIF_ADINR,	0x4,	0x80),
-		RSND_GEN_M_REG(SSI_BUSIF_DALIGN,0x8,	0x80),
-		RSND_GEN_M_REG(SSI_MODE,	0xc,	0x80),
-		RSND_GEN_M_REG(SSI_CTRL,	0x10,	0x80),
-		RSND_GEN_M_REG(SSI_INT_ENABLE,	0x18,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF0_MODE,		0x0,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF0_ADINR,	0x4,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF0_DALIGN,	0x8,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF1_MODE,		0x20,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF1_ADINR,	0x24,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF1_DALIGN,	0x28,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF2_MODE,		0x40,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF2_ADINR,	0x44,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF2_DALIGN,	0x48,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF3_MODE,		0x60,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF3_ADINR,	0x64,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF3_DALIGN,	0x68,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF4_MODE,		0x500,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF4_ADINR,	0x504,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF4_DALIGN,	0x508,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF5_MODE,		0x520,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF5_ADINR,	0x524,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF5_DALIGN,	0x528,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF6_MODE,		0x540,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF6_ADINR,	0x544,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF6_DALIGN,	0x548,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF7_MODE,		0x560,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF7_ADINR,	0x564,	0x80),
+		RSND_GEN_M_REG(SSI_BUSIF7_DALIGN,	0x568,	0x80),
+		RSND_GEN_M_REG(SSI_MODE,		0xc,	0x80),
+		RSND_GEN_M_REG(SSI_CTRL,		0x10,	0x80),
+		RSND_GEN_M_REG(SSI_INT_ENABLE,		0x18,	0x80),
 	};
 
 	static const struct rsnd_regmap_field_conf conf_scu[] = {
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 8f7a0abfa751..4464d1d0a042 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -156,9 +156,30 @@ enum rsnd_reg {
 	RSND_REG_SSI_MODE2,
 	RSND_REG_SSI_CONTROL,
 	RSND_REG_SSI_CTRL,
-	RSND_REG_SSI_BUSIF_MODE,
-	RSND_REG_SSI_BUSIF_ADINR,
-	RSND_REG_SSI_BUSIF_DALIGN,
+	RSND_REG_SSI_BUSIF0_MODE,
+	RSND_REG_SSI_BUSIF0_ADINR,
+	RSND_REG_SSI_BUSIF0_DALIGN,
+	RSND_REG_SSI_BUSIF1_MODE,
+	RSND_REG_SSI_BUSIF1_ADINR,
+	RSND_REG_SSI_BUSIF1_DALIGN,
+	RSND_REG_SSI_BUSIF2_MODE,
+	RSND_REG_SSI_BUSIF2_ADINR,
+	RSND_REG_SSI_BUSIF2_DALIGN,
+	RSND_REG_SSI_BUSIF3_MODE,
+	RSND_REG_SSI_BUSIF3_ADINR,
+	RSND_REG_SSI_BUSIF3_DALIGN,
+	RSND_REG_SSI_BUSIF4_MODE,
+	RSND_REG_SSI_BUSIF4_ADINR,
+	RSND_REG_SSI_BUSIF4_DALIGN,
+	RSND_REG_SSI_BUSIF5_MODE,
+	RSND_REG_SSI_BUSIF5_ADINR,
+	RSND_REG_SSI_BUSIF5_DALIGN,
+	RSND_REG_SSI_BUSIF6_MODE,
+	RSND_REG_SSI_BUSIF6_ADINR,
+	RSND_REG_SSI_BUSIF6_DALIGN,
+	RSND_REG_SSI_BUSIF7_MODE,
+	RSND_REG_SSI_BUSIF7_ADINR,
+	RSND_REG_SSI_BUSIF7_DALIGN,
 	RSND_REG_SSI_INT_ENABLE,
 	RSND_REG_SSI_SYS_STATUS0,
 	RSND_REG_SSI_SYS_STATUS1,
@@ -274,15 +295,12 @@ struct rsnd_mod_ops {
 	int (*fallback)(struct rsnd_mod *mod,
 			struct rsnd_dai_stream *io,
 			struct rsnd_priv *priv);
-	int (*nolock_start)(struct rsnd_mod *mod,
-		    struct rsnd_dai_stream *io,
-		    struct rsnd_priv *priv);
-	int (*nolock_stop)(struct rsnd_mod *mod,
-		    struct rsnd_dai_stream *io,
-		    struct rsnd_priv *priv);
 	int (*prepare)(struct rsnd_mod *mod,
 		       struct rsnd_dai_stream *io,
 		       struct rsnd_priv *priv);
+	int (*cleanup)(struct rsnd_mod *mod,
+		       struct rsnd_dai_stream *io,
+		       struct rsnd_priv *priv);
 };
 
 struct rsnd_dai_stream;
@@ -300,9 +318,8 @@ struct rsnd_mod {
 /*
  * status
  *
- * 0xH0000CBA
+ * 0xH0000CB0
  *
- * A	0: nolock_start	1: nolock_stop
  * B	0: init		1: quit
  * C	0: start	1: stop
  *
@@ -313,9 +330,8 @@ struct rsnd_mod {
  * H	0: hw_params
  * H	0: pointer
  * H	0: prepare
+ * H	0: cleanup
  */
-#define __rsnd_mod_shift_nolock_start	0
-#define __rsnd_mod_shift_nolock_stop	0
 #define __rsnd_mod_shift_init		4
 #define __rsnd_mod_shift_quit		4
 #define __rsnd_mod_shift_start		8
@@ -328,11 +344,12 @@ struct rsnd_mod {
 #define __rsnd_mod_shift_hw_params	28 /* always called */
 #define __rsnd_mod_shift_pointer	28 /* always called */
 #define __rsnd_mod_shift_prepare	28 /* always called */
+#define __rsnd_mod_shift_cleanup	28 /* always called */
 
 #define __rsnd_mod_add_probe		0
 #define __rsnd_mod_add_remove		0
-#define __rsnd_mod_add_nolock_start	 1
-#define __rsnd_mod_add_nolock_stop	-1
+#define __rsnd_mod_add_prepare		0
+#define __rsnd_mod_add_cleanup		0
 #define __rsnd_mod_add_init		 1
 #define __rsnd_mod_add_quit		-1
 #define __rsnd_mod_add_start		 1
@@ -342,10 +359,11 @@ struct rsnd_mod {
 #define __rsnd_mod_add_fallback		0
 #define __rsnd_mod_add_hw_params	0
 #define __rsnd_mod_add_pointer		0
-#define __rsnd_mod_add_prepare		0
 
 #define __rsnd_mod_call_probe		0
 #define __rsnd_mod_call_remove		0
+#define __rsnd_mod_call_prepare		0
+#define __rsnd_mod_call_cleanup		0
 #define __rsnd_mod_call_init		0
 #define __rsnd_mod_call_quit		1
 #define __rsnd_mod_call_start		0
@@ -355,9 +373,6 @@ struct rsnd_mod {
 #define __rsnd_mod_call_fallback	0
 #define __rsnd_mod_call_hw_params	0
 #define __rsnd_mod_call_pointer		0
-#define __rsnd_mod_call_nolock_start	0
-#define __rsnd_mod_call_nolock_stop	1
-#define __rsnd_mod_call_prepare		0
 
 #define rsnd_mod_to_priv(mod)	((mod)->priv)
 #define rsnd_mod_name(mod)	((mod)->ops->name)
@@ -438,6 +453,7 @@ struct rsnd_dai_stream {
 	char name[RSND_DAI_NAME_SIZE];
 	struct snd_pcm_substream *substream;
 	struct rsnd_mod *mod[RSND_MOD_MAX];
+	struct rsnd_mod *dma;
 	struct rsnd_dai *rdai;
 	struct device *dmac_dev; /* for IPMMU */
 	u32 parent_ssi_status;
@@ -467,6 +483,7 @@ struct rsnd_dai {
 
 	int max_channels;	/* 2ch - 16ch */
 	int ssi_lane;		/* 1lane - 4lane */
+	int chan_width;		/* 16/24/32 bit width */
 
 	unsigned int clk_master:1;
 	unsigned int bit_clk_inv:1;
@@ -500,6 +517,11 @@ int rsnd_rdai_channels_ctrl(struct rsnd_dai *rdai,
 int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai,
 			    int ssi_lane);
 
+#define rsnd_rdai_width_set(rdai, width) \
+	rsnd_rdai_width_ctrl(rdai, width)
+#define rsnd_rdai_width_get(rdai) \
+	rsnd_rdai_width_ctrl(rdai, 0)
+int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width);
 void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io);
 int rsnd_dai_connect(struct rsnd_mod *mod,
 		     struct rsnd_dai_stream *io,
@@ -692,6 +714,7 @@ void rsnd_ssi_remove(struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
 int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod);
 int rsnd_ssi_use_busif(struct rsnd_dai_stream *io);
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io);
 u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io);
 
 #define RSND_SSI_HDMI_PORT0	0xf0
@@ -709,7 +732,7 @@ int __rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
 void rsnd_parse_connect_ssi(struct rsnd_dai *rdai,
 			    struct device_node *playback,
 			    struct device_node *capture);
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
 		       int param1, int param2, int *idx);
 
 /*
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index beccfbac7581..cd38a43b976f 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -158,7 +158,7 @@ static int rsnd_src_hw_params(struct rsnd_mod *mod,
 		struct snd_soc_dpcm *dpcm;
 		struct snd_pcm_hw_params *be_params;
 
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+		for_each_dpcm_be(fe, stream, dpcm) {
 			be_params = &dpcm->hw_params;
 
 			if (params_rate(fe_params) != params_rate(be_params))
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 3f880ec66459..fcb4df23248c 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -42,7 +42,13 @@
 #define	DWL_24		(5 << 19)	/* Data Word Length */
 #define	DWL_32		(6 << 19)	/* Data Word Length */
 
+/*
+ * System word length
+ */
+#define	SWL_16		(1 << 16)	/* R/W System Word Length */
+#define	SWL_24		(2 << 16)	/* R/W System Word Length */
 #define	SWL_32		(3 << 16)	/* R/W System Word Length */
+
 #define	SCKD		(1 << 15)	/* Serial Bit Clock Direction */
 #define	SWSD		(1 << 14)	/* Serial WS Direction */
 #define	SCKP		(1 << 13)	/* Serial Bit Clock Polarity */
@@ -72,7 +78,6 @@
 
 struct rsnd_ssi {
 	struct rsnd_mod mod;
-	struct rsnd_mod *dma;
 
 	u32 flags;
 	u32 cr_own;
@@ -145,6 +150,11 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io)
 	return use_busif;
 }
 
+int rsnd_ssi_get_busif(struct rsnd_dai_stream *io)
+{
+	return 0; /* BUSIF0 only for now */
+}
+
 static void rsnd_ssi_status_clear(struct rsnd_mod *mod)
 {
 	rsnd_mod_write(mod, SSISR, 0);
@@ -220,14 +230,32 @@ u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io)
 	return 0;
 }
 
-unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
+static u32 rsnd_rdai_width_to_swl(struct rsnd_dai *rdai)
+{
+	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int width = rsnd_rdai_width_get(rdai);
+
+	switch (width) {
+	case 32: return SWL_32;
+	case 24: return SWL_24;
+	case 16: return SWL_16;
+	}
+
+	dev_err(dev, "unsupported slot width value: %d\n", width);
+	return 0;
+}
+
+unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai,
 		       int param1, int param2, int *idx)
 {
+	struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai);
 	int ssi_clk_mul_table[] = {
 		1, 2, 4, 8, 16, 6, 12,
 	};
 	int j, ret;
 	unsigned int main_rate;
+	int width = rsnd_rdai_width_get(rdai);
 
 	for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) {
 
@@ -240,12 +268,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv,
 		if (j == 0)
 			continue;
 
-		/*
-		 * this driver is assuming that
-		 * system word is 32bit x chan
-		 * see rsnd_ssi_init()
-		 */
-		main_rate = 32 * param1 * param2 * ssi_clk_mul_table[j];
+		main_rate = width * param1 * param2 * ssi_clk_mul_table[j];
 
 		ret = rsnd_adg_clk_query(priv, main_rate);
 		if (ret < 0)
@@ -289,10 +312,15 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 			return -EINVAL;
 		}
 
+		if (ssi->chan != chan) {
+			dev_err(dev, "SSI parent/child should use same chan\n");
+			return -EINVAL;
+		}
+
 		return 0;
 	}
 
-	main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx);
+	main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
 	if (!main_rate) {
 		dev_err(dev, "unsupported clock rate\n");
 		return -EIO;
@@ -312,9 +340,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	 * SSICR  : FORCE, SCKD, SWSD
 	 * SSIWSR : CONT
 	 */
-	ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(idx);
+	ssi->cr_clk = FORCE | rsnd_rdai_width_to_swl(rdai) |
+			SCKD | SWSD | CKDV(idx);
 	ssi->wsr = CONT;
 	ssi->rate = rate;
+	ssi->chan = chan;
 
 	dev_dbg(dev, "%s[%d] outputs %u Hz\n",
 		rsnd_mod_name(mod),
@@ -340,6 +370,7 @@ static void rsnd_ssi_master_clk_stop(struct rsnd_mod *mod,
 
 	ssi->cr_clk	= 0;
 	ssi->rate	= 0;
+	ssi->chan	= 0;
 
 	rsnd_adg_ssi_clk_stop(mod);
 }
@@ -357,15 +388,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 
 	is_tdm = rsnd_runtime_is_ssi_tdm(io);
 
-	/*
-	 * always use 32bit system word.
-	 * see also rsnd_ssi_master_clk_enable()
-	 */
-	cr_own |= FORCE | SWL_32;
+	cr_own |= FORCE | rsnd_rdai_width_to_swl(rdai);
 
 	if (rdai->bit_clk_inv)
 		cr_own |= SCKP;
-	if (rdai->frm_clk_inv ^ is_tdm)
+	if (rdai->frm_clk_inv && !is_tdm)
 		cr_own |= SWSP;
 	if (rdai->data_alignment)
 		cr_own |= SDTA;
@@ -373,6 +400,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_own |= DEL;
 
 	/*
+	 * TDM Mode
+	 * see
+	 *	rsnd_ssiu_init_gen2()
+	 */
+	wsr = ssi->wsr;
+	if (is_tdm) {
+		wsr	|= WS_MODE;
+		cr_own	|= CHNL_8;
+	}
+
+	/*
 	 * We shouldn't exchange SWSP after running.
 	 * This means, parent needs to care it.
 	 */
@@ -384,6 +422,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 
 	cr_own &= ~DWL_MASK;
 	switch (snd_pcm_format_width(runtime->format)) {
+	case 8:
+		cr_own |= DWL_8;
+		break;
 	case 16:
 		cr_own |= DWL_16;
 		break;
@@ -399,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 		cr_mode = DIEN;		/* PIO : enable Data interrupt */
 	}
 
-	/*
-	 * TDM Extend Mode
-	 * see
-	 *	rsnd_ssiu_init_gen2()
-	 */
-	wsr = ssi->wsr;
-	if (is_tdm) {
-		wsr	|= WS_MODE;
-		cr_own	|= CHNL_8;
-	}
 init_end:
 	ssi->cr_own	= cr_own;
 	ssi->cr_mode	= cr_mode;
@@ -488,26 +519,16 @@ static int rsnd_ssi_hw_params(struct rsnd_mod *mod,
 			      struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
-	int chan = params_channels(params);
+	struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+	unsigned int fmt_width = snd_pcm_format_width(params_format(params));
 
-	/*
-	 * snd_pcm_ops::hw_params will be called *before*
-	 * snd_soc_dai_ops::trigger. Thus, ssi->usrcnt is 0
-	 * in 1st call.
-	 */
-	if (ssi->usrcnt) {
-		/*
-		 * Already working.
-		 * It will happen if SSI has parent/child connection.
-		 * it is error if child <-> parent SSI uses
-		 * different channels.
-		 */
-		if (ssi->chan != chan)
-			return -EIO;
-	}
+	if (fmt_width > rdai->chan_width) {
+		struct rsnd_priv *priv = rsnd_io_to_priv(io);
+		struct device *dev = rsnd_priv_to_dev(priv);
 
-	ssi->chan = chan;
+		dev_err(dev, "invalid combination of slot-width and format-data-width\n");
+		return -EINVAL;
+	}
 
 	return 0;
 }
@@ -873,7 +894,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 			      struct rsnd_dai_stream *io,
 			      struct rsnd_priv *priv)
 {
-	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
 	int ret;
 
 	/*
@@ -888,7 +908,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
 		return ret;
 
 	/* SSI probe might be called many times in MUX multi path */
-	ret = rsnd_dma_attach(io, mod, &ssi->dma);
+	ret = rsnd_dma_attach(io, mod, &io->dma);
 
 	return ret;
 }
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index 016fbf5ac242..39b67643b5dc 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -10,9 +10,12 @@
 
 struct rsnd_ssiu {
 	struct rsnd_mod mod;
+	u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */
+	unsigned int usrcnt;
 };
 
 #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr)
+#define rsnd_mod_to_ssiu(_mod) container_of((_mod), struct rsnd_ssiu, mod)
 #define for_each_rsnd_ssiu(pos, priv, i)				\
 	for (i = 0;							\
 	     (i < rsnd_ssiu_nr(priv)) &&				\
@@ -120,6 +123,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 			       struct rsnd_dai_stream *io,
 			       struct rsnd_priv *priv)
 {
+	struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
 	int hdmi = rsnd_ssi_hdmi_port(io);
 	int ret;
 	u32 mode = 0;
@@ -128,6 +132,8 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 	if (ret < 0)
 		return ret;
 
+	ssiu->usrcnt++;
+
 	if (rsnd_runtime_is_ssi_tdm(io)) {
 		/*
 		 * TDM Extend Mode
@@ -140,15 +146,59 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 	rsnd_mod_write(mod, SSI_MODE, mode);
 
 	if (rsnd_ssi_use_busif(io)) {
-		rsnd_mod_write(mod, SSI_BUSIF_ADINR,
-			       rsnd_get_adinr_bit(mod, io) |
-			       (rsnd_io_is_play(io) ?
-				rsnd_runtime_channel_after_ctu(io) :
-				rsnd_runtime_channel_original(io)));
-		rsnd_mod_write(mod, SSI_BUSIF_MODE,
-			       rsnd_get_busif_shift(io, mod) | 1);
-		rsnd_mod_write(mod, SSI_BUSIF_DALIGN,
-			       rsnd_get_dalign(mod, io));
+		int id = rsnd_mod_id(mod);
+		int busif = rsnd_ssi_get_busif(io);
+
+		/*
+		 * FIXME
+		 *
+		 * We can't support SSI9-4/5/6/7, because its address is
+		 * out of calculation rule
+		 */
+		if ((id == 9) && (busif >= 4)) {
+			struct device *dev = rsnd_priv_to_dev(priv);
+
+			dev_err(dev, "This driver doesn't support SSI%d-%d, so far",
+				id, busif);
+		}
+
+#define RSND_WRITE_BUSIF(i)						\
+		rsnd_mod_write(mod, SSI_BUSIF##i##_ADINR,		\
+			       rsnd_get_adinr_bit(mod, io) |		\
+			       (rsnd_io_is_play(io) ?			\
+				rsnd_runtime_channel_after_ctu(io) :	\
+				rsnd_runtime_channel_original(io)));	\
+		rsnd_mod_write(mod, SSI_BUSIF##i##_MODE,		\
+			       rsnd_get_busif_shift(io, mod) | 1);	\
+		rsnd_mod_write(mod, SSI_BUSIF##i##_DALIGN,		\
+			       rsnd_get_dalign(mod, io))
+
+		switch (busif) {
+		case 0:
+			RSND_WRITE_BUSIF(0);
+			break;
+		case 1:
+			RSND_WRITE_BUSIF(1);
+			break;
+		case 2:
+			RSND_WRITE_BUSIF(2);
+			break;
+		case 3:
+			RSND_WRITE_BUSIF(3);
+			break;
+		case 4:
+			RSND_WRITE_BUSIF(4);
+			break;
+		case 5:
+			RSND_WRITE_BUSIF(5);
+			break;
+		case 6:
+			RSND_WRITE_BUSIF(6);
+			break;
+		case 7:
+			RSND_WRITE_BUSIF(7);
+			break;
+		}
 	}
 
 	if (hdmi) {
@@ -194,10 +244,12 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod,
 				struct rsnd_dai_stream *io,
 				struct rsnd_priv *priv)
 {
+	int busif = rsnd_ssi_get_busif(io);
+
 	if (!rsnd_ssi_use_busif(io))
 		return 0;
 
-	rsnd_mod_write(mod, SSI_CTRL, 0x1);
+	rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 1 << (busif * 4));
 
 	if (rsnd_ssi_multi_slaves_runtime(io))
 		rsnd_mod_write(mod, SSI_CONTROL, 0x1);
@@ -209,10 +261,16 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod,
 			       struct rsnd_dai_stream *io,
 			       struct rsnd_priv *priv)
 {
+	struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+	int busif = rsnd_ssi_get_busif(io);
+
 	if (!rsnd_ssi_use_busif(io))
 		return 0;
 
-	rsnd_mod_write(mod, SSI_CTRL, 0);
+	rsnd_mod_bset(mod, SSI_CTRL, 1 << (busif * 4), 0);
+
+	if (--ssiu->usrcnt)
+		return 0;
 
 	if (rsnd_ssi_multi_slaves_runtime(io))
 		rsnd_mod_write(mod, SSI_CONTROL, 0);
@@ -246,6 +304,16 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io,
 	return rsnd_dai_connect(mod, io, mod->type);
 }
 
+static u32 *rsnd_ssiu_get_status(struct rsnd_dai_stream *io,
+				 struct rsnd_mod *mod,
+				 enum rsnd_mod_type type)
+{
+	struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod);
+	int busif = rsnd_ssi_get_busif(io);
+
+	return &ssiu->busif_status[busif];
+}
+
 int rsnd_ssiu_probe(struct rsnd_priv *priv)
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
@@ -269,7 +337,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv)
 
 	for_each_rsnd_ssiu(ssiu, priv, i) {
 		ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu),
-				    ops, NULL, rsnd_mod_get_status,
+				    ops, NULL, rsnd_ssiu_get_status,
 				    RSND_MOD_SSIU, i);
 		if (ret)
 			return ret;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 409d082e80d1..699397a09167 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -157,7 +157,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
 	ret = dpcm_be_dai_startup(fe, stream);
 	if (ret < 0) {
 		/* clean up all links */
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		for_each_dpcm_be(fe, stream, dpcm)
 			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 		dpcm_be_disconnect(fe, stream);
@@ -321,7 +321,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream)
 	ret = dpcm_be_dai_shutdown(fe, stream);
 
 	/* mark FE's links ready to prune */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+	for_each_dpcm_be(fe, stream, dpcm)
 		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 	dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 473eefe8658e..6ddcf12bc030 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -52,6 +52,10 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root);
 
 static DEFINE_MUTEX(client_mutex);
 static LIST_HEAD(component_list);
+static LIST_HEAD(unbind_card_list);
+
+#define for_each_component(component)			\
+	list_for_each_entry(component, &component_list, list)
 
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -62,8 +66,9 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
-/* If a DMI filed contain strings in this blacklist (e.g.
- * "Type2 - Board Manufacturer" or  "Type1 - TBD by OEM"), it will be taken
+/*
+ * If a DMI filed contain strings in this blacklist (e.g.
+ * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken
  * as invalid and dropped when setting the card long name from DMI info.
  */
 static const char * const dmi_blacklist[] = {
@@ -175,8 +180,8 @@ static int dai_list_show(struct seq_file *m, void *v)
 
 	mutex_lock(&client_mutex);
 
-	list_for_each_entry(component, &component_list, list)
-		list_for_each_entry(dai, &component->dai_list, list)
+	for_each_component(component)
+		for_each_component_dais(component, dai)
 			seq_printf(m, "%s\n", dai->name);
 
 	mutex_unlock(&client_mutex);
@@ -191,7 +196,7 @@ static int component_list_show(struct seq_file *m, void *v)
 
 	mutex_lock(&client_mutex);
 
-	list_for_each_entry(component, &component_list, list)
+	for_each_component(component)
 		seq_printf(m, "%s\n", component->name);
 
 	mutex_unlock(&client_mutex);
@@ -218,7 +223,7 @@ static void soc_init_card_debugfs(struct snd_soc_card *card)
 						    &card->pop_time);
 	if (!card->debugfs_pop_time)
 		dev_warn(card->dev,
-		       "ASoC: Failed to create pop time debugfs file\n");
+			 "ASoC: Failed to create pop time debugfs file\n");
 }
 
 static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
@@ -341,7 +346,7 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->dai_link->no_pcm &&
 			!strcmp(rtd->dai_link->name, dai_link))
 			return rtd->pcm->streams[stream].substream;
@@ -398,7 +403,7 @@ static void soc_remove_pcm_runtimes(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd, *_rtd;
 
-	list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) {
+	for_each_card_rtds_safe(card, rtd, _rtd) {
 		list_del(&rtd->list);
 		soc_free_pcm_runtime(rtd);
 	}
@@ -411,7 +416,7 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (!strcmp(rtd->dai_link->name, dai_link))
 			return rtd;
 	}
@@ -422,7 +427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
 
 static void codec2codec_close_delayed_work(struct work_struct *work)
 {
-	/* Currently nothing to do for c2c links
+	/*
+	 * Currently nothing to do for c2c links
 	 * Since c2c links are internal nodes in the DAPM graph and
 	 * don't interface with the outside world or application layer
 	 * we don't have to do any special handling on close.
@@ -442,8 +448,9 @@ int snd_soc_suspend(struct device *dev)
 	if (!card->instantiated)
 		return 0;
 
-	/* Due to the resume being scheduled into a workqueue we could
-	* suspend before that's finished - wait for it to complete.
+	/*
+	 * Due to the resume being scheduled into a workqueue we could
+	 * suspend before that's finished - wait for it to complete.
 	 */
 	snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0);
 
@@ -451,13 +458,13 @@ int snd_soc_suspend(struct device *dev)
 	snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot);
 
 	/* mute any active DACs */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *dai;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			struct snd_soc_dai *dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, dai) {
 			struct snd_soc_dai_driver *drv = dai->driver;
 
 			if (drv->ops->digital_mute && dai->playback_active)
@@ -466,7 +473,7 @@ int snd_soc_suspend(struct device *dev)
 	}
 
 	/* suspend all pcms */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
@@ -476,7 +483,7 @@ int snd_soc_suspend(struct device *dev)
 	if (card->suspend_pre)
 		card->suspend_pre(card);
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -487,10 +494,10 @@ int snd_soc_suspend(struct device *dev)
 	}
 
 	/* close any waiting streams */
-	list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_card_rtds(card, rtd)
 		flush_delayed_work(&rtd->delayed_work);
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
@@ -509,11 +516,14 @@ int snd_soc_suspend(struct device *dev)
 	snd_soc_dapm_sync(&card->dapm);
 
 	/* suspend all COMPONENTs */
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
-		struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	for_each_card_components(card, component) {
+		struct snd_soc_dapm_context *dapm =
+				snd_soc_component_get_dapm(component);
 
-		/* If there are paths active then the COMPONENT will be held with
-		 * bias _ON and should not be suspended. */
+		/*
+		 * If there are paths active then the COMPONENT will be held
+		 * with bias _ON and should not be suspended.
+		 */
 		if (!component->suspended) {
 			switch (snd_soc_dapm_get_bias_level(dapm)) {
 			case SND_SOC_BIAS_STANDBY:
@@ -547,7 +557,7 @@ int snd_soc_suspend(struct device *dev)
 		}
 	}
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -567,18 +577,21 @@ int snd_soc_suspend(struct device *dev)
 }
 EXPORT_SYMBOL_GPL(snd_soc_suspend);
 
-/* deferred resume work, so resume can complete before we finished
+/*
+ * deferred resume work, so resume can complete before we finished
  * setting our codec back up, which can be very slow on I2C
  */
 static void soc_resume_deferred(struct work_struct *work)
 {
 	struct snd_soc_card *card =
-			container_of(work, struct snd_soc_card, deferred_resume_work);
+			container_of(work, struct snd_soc_card,
+				     deferred_resume_work);
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_component *component;
 	int i;
 
-	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
+	/*
+	 * our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
 	 * so userspace apps are blocked from touching us
 	 */
 
@@ -591,7 +604,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		card->resume_pre(card);
 
 	/* resume control bus DAIs */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -601,7 +614,7 @@ static void soc_resume_deferred(struct work_struct *work)
 			cpu_dai->driver->resume(cpu_dai);
 	}
 
-	list_for_each_entry(component, &card->component_dev_list, card_list) {
+	for_each_card_components(card, component) {
 		if (component->suspended) {
 			if (component->driver->resume)
 				component->driver->resume(component);
@@ -609,7 +622,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		}
 	}
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
@@ -624,13 +637,13 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	/* unmute any active DACs */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *dai;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			struct snd_soc_dai *dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, dai) {
 			struct snd_soc_dai_driver *drv = dai->driver;
 
 			if (drv->ops->digital_mute && dai->playback_active)
@@ -638,7 +651,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		}
 	}
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 		if (rtd->dai_link->ignore_suspend)
@@ -673,16 +686,15 @@ int snd_soc_resume(struct device *dev)
 		return 0;
 
 	/* activate pins from sleep state */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
-		struct snd_soc_dai **codec_dais = rtd->codec_dais;
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *codec_dai;
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 		int j;
 
 		if (cpu_dai->active)
 			pinctrl_pm_select_default_state(cpu_dai->dev);
 
-		for (j = 0; j < rtd->num_codecs; j++) {
-			struct snd_soc_dai *codec_dai = codec_dais[j];
+		for_each_rtd_codec_dai(rtd, j, codec_dai) {
 			if (codec_dai->active)
 				pinctrl_pm_select_default_state(codec_dai->dev);
 		}
@@ -694,8 +706,9 @@ int snd_soc_resume(struct device *dev)
 	 * have that problem and may take a substantial amount of time to resume
 	 * due to I/O costs and anti-pop so handle them out of line.
 	 */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
 		bus_control |= cpu_dai->driver->bus_control;
 	}
 	if (bus_control) {
@@ -725,7 +738,7 @@ static struct snd_soc_component *soc_find_component(
 
 	lockdep_assert_held(&client_mutex);
 
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 		if (of_node) {
 			if (component->dev->of_node == of_node)
 				return component;
@@ -737,6 +750,24 @@ static struct snd_soc_component *soc_find_component(
 	return NULL;
 }
 
+static int snd_soc_is_matching_component(
+	const struct snd_soc_dai_link_component *dlc,
+	struct snd_soc_component *component)
+{
+	struct device_node *component_of_node;
+
+	component_of_node = component->dev->of_node;
+	if (!component_of_node && component->dev->parent)
+		component_of_node = component->dev->parent->of_node;
+
+	if (dlc->of_node && component_of_node != dlc->of_node)
+		return 0;
+	if (dlc->name && strcmp(component->name, dlc->name))
+		return 0;
+
+	return 1;
+}
+
 /**
  * snd_soc_find_dai - Find a registered DAI
  *
@@ -753,21 +784,14 @@ struct snd_soc_dai *snd_soc_find_dai(
 {
 	struct snd_soc_component *component;
 	struct snd_soc_dai *dai;
-	struct device_node *component_of_node;
 
 	lockdep_assert_held(&client_mutex);
 
-	/* Find CPU DAI from registered DAIs*/
-	list_for_each_entry(component, &component_list, list) {
-		component_of_node = component->dev->of_node;
-		if (!component_of_node && component->dev->parent)
-			component_of_node = component->dev->parent->of_node;
-
-		if (dlc->of_node && component_of_node != dlc->of_node)
-			continue;
-		if (dlc->name && strcmp(component->name, dlc->name))
+	/* Find CPU DAI from registered DAIs */
+	for_each_component(component) {
+		if (!snd_soc_is_matching_component(dlc, component))
 			continue;
-		list_for_each_entry(dai, &component->dai_list, list) {
+		for_each_component_dais(component, dai) {
 			if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)
 			    && (!dai->driver->name
 				|| strcmp(dai->driver->name, dlc->dai_name)))
@@ -781,7 +805,6 @@ struct snd_soc_dai *snd_soc_find_dai(
 }
 EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
-
 /**
  * snd_soc_find_dai_link - Find a DAI link
  *
@@ -805,7 +828,7 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card,
 
 	lockdep_assert_held(&client_mutex);
 
-	list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+	for_each_card_links_safe(card, link, _link) {
 		if (link->id != id)
 			continue;
 
@@ -828,7 +851,7 @@ static bool soc_is_dai_link_bound(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		if (rtd->dai_link == dai_link)
 			return true;
 	}
@@ -844,8 +867,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	struct snd_soc_dai_link_component cpu_dai_component;
 	struct snd_soc_component *component;
 	struct snd_soc_dai **codec_dais;
-	struct device_node *platform_of_node;
-	const char *platform_name;
 	int i;
 
 	if (dai_link->ignore)
@@ -877,6 +898,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	rtd->num_codecs = dai_link->num_codecs;
 
 	/* Find CODEC from registered CODECs */
+	/* we can use for_each_rtd_codec_dai() after this */
 	codec_dais = rtd->codec_dais;
 	for (i = 0; i < rtd->num_codecs; i++) {
 		codec_dais[i] = snd_soc_find_dai(&codecs[i]);
@@ -891,24 +913,11 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 	/* Single codec links expect codec and codec_dai in runtime data */
 	rtd->codec_dai = codec_dais[0];
 
-	/* if there's no platform we match on the empty platform */
-	platform_name = dai_link->platform_name;
-	if (!platform_name && !dai_link->platform_of_node)
-		platform_name = "snd-soc-dummy";
-
 	/* find one from the set of registered platforms */
-	list_for_each_entry(component, &component_list, list) {
-		platform_of_node = component->dev->of_node;
-		if (!platform_of_node && component->dev->parent->of_node)
-			platform_of_node = component->dev->parent->of_node;
-
-		if (dai_link->platform_of_node) {
-			if (platform_of_node != dai_link->platform_of_node)
-				continue;
-		} else {
-			if (strcmp(component->name, platform_name))
-				continue;
-		}
+	for_each_component(component) {
+		if (!snd_soc_is_matching_component(dai_link->platform,
+						   component))
+			continue;
 
 		snd_soc_rtdcom_add(rtd, component);
 	}
@@ -918,7 +927,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card,
 
 _err_defer:
 	soc_free_pcm_runtime(rtd);
-	return  -EPROBE_DEFER;
+	return -EPROBE_DEFER;
 }
 
 static void soc_remove_component(struct snd_soc_component *component)
@@ -942,23 +951,25 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order)
 {
 	int err;
 
-	if (dai && dai->probed &&
-			dai->driver->remove_order == order) {
-		if (dai->driver->remove) {
-			err = dai->driver->remove(dai);
-			if (err < 0)
-				dev_err(dai->dev,
-					"ASoC: failed to remove %s: %d\n",
-					dai->name, err);
-		}
-		dai->probed = 0;
+	if (!dai || !dai->probed ||
+	    dai->driver->remove_order != order)
+		return;
+
+	if (dai->driver->remove) {
+		err = dai->driver->remove(dai);
+		if (err < 0)
+			dev_err(dai->dev,
+				"ASoC: failed to remove %s: %d\n",
+				dai->name, err);
 	}
+	dai->probed = 0;
 }
 
 static void soc_remove_link_dais(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd, int order)
 {
 	int i;
+	struct snd_soc_dai *codec_dai;
 
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
@@ -967,8 +978,8 @@ static void soc_remove_link_dais(struct snd_soc_card *card,
 	}
 
 	/* remove the CODEC DAI */
-	for (i = 0; i < rtd->num_codecs; i++)
-		soc_remove_dai(rtd->codec_dais[i], order);
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		soc_remove_dai(codec_dai, order);
 
 	soc_remove_dai(rtd->cpu_dai, order);
 }
@@ -993,28 +1004,57 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai_link *link, *_link;
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd)
 			soc_remove_link_dais(card, rtd, order);
 	}
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd)
 			soc_remove_link_components(card, rtd, order);
 	}
 
-	list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+	for_each_card_links_safe(card, link, _link) {
 		if (link->dobj.type == SND_SOC_DOBJ_DAI_LINK)
 			dev_warn(card->dev, "Topology forgot to remove link %s?\n",
 				link->name);
 
 		list_del(&link->list);
-		card->num_dai_links--;
 	}
 }
 
+static int snd_soc_init_platform(struct snd_soc_card *card,
+				 struct snd_soc_dai_link *dai_link)
+{
+	struct snd_soc_dai_link_component *platform = dai_link->platform;
+
+	/*
+	 * FIXME
+	 *
+	 * this function should be removed in the future
+	 */
+	/* convert Legacy platform link */
+	if (!platform) {
+		platform = devm_kzalloc(card->dev,
+				sizeof(struct snd_soc_dai_link_component),
+				GFP_KERNEL);
+		if (!platform)
+			return -ENOMEM;
+
+		dai_link->platform	= platform;
+		platform->name		= dai_link->platform_name;
+		platform->of_node	= dai_link->platform_of_node;
+		platform->dai_name	= NULL;
+	}
+
+	/* if there's no platform we match on the empty platform */
+	if (!platform->name &&
+	    !platform->of_node)
+		platform->name = "snd-soc-dummy";
+
+	return 0;
+}
+
 static int snd_soc_init_multicodec(struct snd_soc_card *card,
 				   struct snd_soc_dai_link *dai_link)
 {
@@ -1043,9 +1083,16 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card,
 }
 
 static int soc_init_dai_link(struct snd_soc_card *card,
-				   struct snd_soc_dai_link *link)
+			     struct snd_soc_dai_link *link)
 {
 	int i, ret;
+	struct snd_soc_dai_link_component *codec;
+
+	ret = snd_soc_init_platform(card, link);
+	if (ret) {
+		dev_err(card->dev, "ASoC: failed to init multiplatform\n");
+		return ret;
+	}
 
 	ret = snd_soc_init_multicodec(card, link);
 	if (ret) {
@@ -1053,19 +1100,19 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 		return ret;
 	}
 
-	for (i = 0; i < link->num_codecs; i++) {
+	for_each_link_codecs(link, i, codec) {
 		/*
 		 * Codec must be specified by 1 of name or OF node,
 		 * not both or neither.
 		 */
-		if (!!link->codecs[i].name ==
-		    !!link->codecs[i].of_node) {
+		if (!!codec->name ==
+		    !!codec->of_node) {
 			dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n",
 				link->name);
 			return -EINVAL;
 		}
 		/* Codec DAI name must be specified */
-		if (!link->codecs[i].dai_name) {
+		if (!codec->dai_name) {
 			dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n",
 				link->name);
 			return -EINVAL;
@@ -1076,13 +1123,12 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 	 * Platform may be specified by either name or OF node, but
 	 * can be left unspecified, and a dummy platform will be used.
 	 */
-	if (link->platform_name && link->platform_of_node) {
+	if (link->platform->name && link->platform->of_node) {
 		dev_err(card->dev,
 			"ASoC: Both platform name/of_node are set for %s\n",
 			link->name);
 		return -EINVAL;
 	}
-
 	/*
 	 * CPU device may be specified by either name or OF node, but
 	 * can be left unspecified, and will be matched based on DAI
@@ -1111,7 +1157,8 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 
 void snd_soc_disconnect_sync(struct device *dev)
 {
-	struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL);
+	struct snd_soc_component *component =
+			snd_soc_lookup_component(dev, NULL);
 
 	if (!component || !component->card)
 		return;
@@ -1142,14 +1189,14 @@ int snd_soc_add_dai_link(struct snd_soc_card *card,
 	}
 
 	lockdep_assert_held(&client_mutex);
-	/* Notify the machine driver for extra initialization
+	/*
+	 * Notify the machine driver for extra initialization
 	 * on the link created by topology.
 	 */
 	if (dai_link->dobj.type && card->add_dai_link)
 		card->add_dai_link(card, dai_link);
 
 	list_add_tail(&dai_link->list, &card->dai_link_list);
-	card->num_dai_links++;
 
 	return 0;
 }
@@ -1178,16 +1225,16 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 	}
 
 	lockdep_assert_held(&client_mutex);
-	/* Notify the machine driver for extra destruction
+	/*
+	 * Notify the machine driver for extra destruction
 	 * on the link created by topology.
 	 */
 	if (dai_link->dobj.type && card->remove_dai_link)
 		card->remove_dai_link(card, dai_link);
 
-	list_for_each_entry_safe(link, _link, &card->dai_link_list, list) {
+	for_each_card_links_safe(card, link, _link) {
 		if (link == dai_link) {
 			list_del(&link->list);
-			card->num_dai_links--;
 			return;
 		}
 	}
@@ -1239,7 +1286,8 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
 static int soc_probe_component(struct snd_soc_card *card,
 	struct snd_soc_component *component)
 {
-	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
 	struct snd_soc_dai *dai;
 	int ret;
 
@@ -1277,7 +1325,7 @@ static int soc_probe_component(struct snd_soc_card *card,
 		}
 	}
 
-	list_for_each_entry(dai, &component->dai_list, list) {
+	for_each_component_dais(component, dai) {
 		ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
 		if (ret != 0) {
 			dev_err(component->dev,
@@ -1320,6 +1368,7 @@ static int soc_probe_component(struct snd_soc_card *card,
 					component->driver->num_dapm_routes);
 
 	list_add(&dapm->list, &card->dapm_list);
+	/* see for_each_card_components */
 	list_add(&component->card_list, &card->component_dev_list);
 
 	return 0;
@@ -1370,8 +1419,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
 }
 
 static int soc_probe_link_components(struct snd_soc_card *card,
-			struct snd_soc_pcm_runtime *rtd,
-				     int order)
+				     struct snd_soc_pcm_runtime *rtd, int order)
 {
 	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
@@ -1398,6 +1446,7 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order)
 
 	if (dai->driver->probe) {
 		int ret = dai->driver->probe(dai);
+
 		if (ret < 0) {
 			dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n",
 				dai->name, ret);
@@ -1431,48 +1480,6 @@ static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
 	return 0;
 }
 
-static int soc_link_dai_widgets(struct snd_soc_card *card,
-				struct snd_soc_dai_link *dai_link,
-				struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dapm_widget *sink, *source;
-	int ret;
-
-	if (rtd->num_codecs > 1)
-		dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n");
-
-	/* link the DAI widgets */
-	sink = codec_dai->playback_widget;
-	source = cpu_dai->capture_widget;
-	if (sink && source) {
-		ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
-					   dai_link->num_params,
-					   source, sink);
-		if (ret != 0) {
-			dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
-				sink->name, source->name, ret);
-			return ret;
-		}
-	}
-
-	sink = cpu_dai->playback_widget;
-	source = codec_dai->capture_widget;
-	if (sink && source) {
-		ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params,
-					   dai_link->num_params,
-					   source, sink);
-		if (ret != 0) {
-			dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n",
-				sink->name, source->name, ret);
-			return ret;
-		}
-	}
-
-	return 0;
-}
-
 static int soc_probe_link_dais(struct snd_soc_card *card,
 		struct snd_soc_pcm_runtime *rtd, int order)
 {
@@ -1480,6 +1487,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_component *component;
+	struct snd_soc_dai *codec_dai;
 	int i, ret, num;
 
 	dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n",
@@ -1493,8 +1501,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		return ret;
 
 	/* probe the CODEC DAI */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		ret = soc_probe_dai(rtd->codec_dais[i], order);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		ret = soc_probe_dai(codec_dai, order);
 		if (ret)
 			return ret;
 	}
@@ -1546,7 +1554,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 	}
 
 	if (cpu_dai->driver->compress_new) {
-		/*create compress_device"*/
+		/* create compress_device" */
 		ret = cpu_dai->driver->compress_new(rtd, num);
 		if (ret < 0) {
 			dev_err(card->dev, "ASoC: can't create compress %s\n",
@@ -1560,7 +1568,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 			ret = soc_new_pcm(rtd, num);
 			if (ret < 0) {
 				dev_err(card->dev, "ASoC: can't create pcm %s :%d\n",
-				       dai_link->stream_name, ret);
+					dai_link->stream_name, ret);
 				return ret;
 			}
 			ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd);
@@ -1573,11 +1581,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		} else {
 			INIT_DELAYED_WORK(&rtd->delayed_work,
 						codec2codec_close_delayed_work);
-
-			/* link the DAI widgets */
-			ret = soc_link_dai_widgets(card, dai_link, rtd);
-			if (ret)
-				return ret;
 		}
 	}
 
@@ -1628,8 +1631,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card)
 	int order;
 	int ret;
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-		order++) {
+	for_each_comp_order(order) {
 		list_for_each_entry(comp, &card->aux_comp_list, card_aux_list) {
 			if (comp->driver->probe_order == order) {
 				ret = soc_probe_component(card,	comp);
@@ -1651,8 +1653,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 	struct snd_soc_component *comp, *_comp;
 	int order;
 
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-		order++) {
+	for_each_comp_order(order) {
 		list_for_each_entry_safe(comp, _comp,
 			&card->aux_comp_list, card_aux_list) {
 
@@ -1681,14 +1682,12 @@ static void soc_remove_aux_devices(struct snd_soc_card *card)
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt)
 {
-	struct snd_soc_dai **codec_dais = rtd->codec_dais;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	unsigned int i;
 	int ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = codec_dais[i];
-
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
 		if (ret != 0 && ret != -ENOTSUPP) {
 			dev_warn(codec_dai->dev,
@@ -1697,8 +1696,10 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 		}
 	}
 
-	/* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */
-	/* the component which has non_legacy_dai_naming is Codec */
+	/*
+	 * Flip the polarity for the "CPU" end of a CODEC<->CODEC link
+	 * the component which has non_legacy_dai_naming is Codec
+	 */
 	if (cpu_dai->component->driver->non_legacy_dai_naming) {
 		unsigned int inv_dai_fmt;
 
@@ -1732,9 +1733,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 }
 EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt);
 
-
 #ifdef CONFIG_DMI
-/* Trim special characters, and replace '-' with '_' since '-' is used to
+/*
+ * Trim special characters, and replace '-' with '_' since '-' is used to
  * separate different DMI fields in the card long name. Only number and
  * alphabet characters and a few separator characters are kept.
  */
@@ -1753,7 +1754,8 @@ static void cleanup_dmi_name(char *name)
 	name[j] = '\0';
 }
 
-/* Check if a DMI field is valid, i.e. not containing any string
+/*
+ * Check if a DMI field is valid, i.e. not containing any string
  * in the black list.
  */
 static int is_dmi_valid(const char *field)
@@ -1816,7 +1818,6 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 		return 0;
 	}
 
-
 	snprintf(card->dmi_longname, sizeof(card->snd_card->longname),
 			 "%s", vendor);
 	cleanup_dmi_name(card->dmi_longname);
@@ -1832,7 +1833,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour)
 		if (len < longname_buf_size)
 			cleanup_dmi_name(card->dmi_longname + len);
 
-		/* some vendors like Lenovo may only put a self-explanatory
+		/*
+		 * some vendors like Lenovo may only put a self-explanatory
 		 * name in the product version field
 		 */
 		product_version = dmi_get_system_info(DMI_PRODUCT_VERSION);
@@ -1891,7 +1893,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 	struct snd_soc_dai_link *dai_link;
 	int i;
 
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 
 		/* does this component override FEs ? */
 		if (!component->driver->ignore_machine)
@@ -1903,9 +1905,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 			continue;
 
 		/* machine matches, so override the rtd data */
-		for (i = 0; i < card->num_links; i++) {
-
-			dai_link = &card->dai_link[i];
+		for_each_card_prelinks(card, i, dai_link) {
 
 			/* ignore this FE */
 			if (dai_link->dynamic) {
@@ -1917,7 +1917,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 				 card->dai_link[i].name);
 
 			/* override platform component */
-			dai_link->platform_name = component->name;
+			if (snd_soc_init_platform(card, dai_link) < 0) {
+				dev_err(card->dev, "init platform error");
+				continue;
+			}
+			dai_link->platform->name = component->name;
 
 			/* convert non BE into BE */
 			dai_link->no_pcm = 1;
@@ -1926,7 +1930,8 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 			dai_link->be_hw_params_fixup =
 				component->driver->be_hw_params_fixup;
 
-			/* most BE links don't set stream name, so set it to
+			/*
+			 * most BE links don't set stream name, so set it to
 			 * dai link name if it's NULL to help bind widgets.
 			 */
 			if (!dai_link->stream_name)
@@ -1936,7 +1941,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card)
 		/* Inform userspace we are using alternate topology */
 		if (component->driver->topology_name_prefix) {
 
-			/* topology shortname created ? */
+			/* topology shortname created? */
 			if (!card->topology_shortname_created) {
 				comp_drv = component->driver;
 
@@ -1965,8 +1970,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	soc_check_tplg_fes(card);
 
 	/* bind DAIs */
-	for (i = 0; i < card->num_links; i++) {
-		ret = soc_bind_dai_link(card, &card->dai_link[i]);
+	for_each_card_prelinks(card, i, dai_link) {
+		ret = soc_bind_dai_link(card, dai_link);
 		if (ret != 0)
 			goto base_error;
 	}
@@ -1979,8 +1984,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* add predefined DAI links to the list */
-	for (i = 0; i < card->num_links; i++)
-		snd_soc_add_dai_link(card, card->dai_link+i);
+	for_each_card_prelinks(card, i, dai_link)
+		snd_soc_add_dai_link(card, dai_link);
 
 	/* card bind complete so register a sound card */
 	ret = snd_card_new(card->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
@@ -2024,9 +2029,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* probe all components used by DAI links on this card */
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
 			ret = soc_probe_link_components(card, rtd, order);
 			if (ret < 0) {
 				dev_err(card->dev,
@@ -2042,10 +2046,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	if (ret < 0)
 		goto probe_dai_err;
 
-	/* Find new DAI links added during probing components and bind them.
+	/*
+	 * Find new DAI links added during probing components and bind them.
 	 * Components with topology may bring new DAIs and DAI links.
 	 */
-	list_for_each_entry(dai_link, &card->dai_link_list, list) {
+	for_each_card_links(card, dai_link) {
 		if (soc_is_dai_link_bound(card, dai_link))
 			continue;
 
@@ -2058,9 +2063,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	/* probe all DAI links on this card */
-	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
-			order++) {
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_comp_order(order) {
+		for_each_card_rtds(card, rtd) {
 			ret = soc_probe_link_dais(card, rtd, order);
 			if (ret < 0) {
 				dev_err(card->dev,
@@ -2075,7 +2079,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	snd_soc_dapm_connect_dai_link_widgets(card);
 
 	if (card->controls)
-		snd_soc_add_card_controls(card, card->controls, card->num_controls);
+		snd_soc_add_card_controls(card, card->controls,
+					  card->num_controls);
 
 	if (card->dapm_routes)
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
@@ -2181,7 +2186,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 
 	/* make sure any delayed work runs */
-	list_for_each_entry(rtd, &card->rtd_list, list)
+	for_each_card_rtds(card, rtd)
 		flush_delayed_work(&rtd->delayed_work);
 
 	/* free the ALSA card at first; this syncs with pending operations */
@@ -2221,21 +2226,23 @@ int snd_soc_poweroff(struct device *dev)
 	if (!card->instantiated)
 		return 0;
 
-	/* Flush out pmdown_time work - we actually do want to run it
-	 * now, we're shutting down so no imminent restart. */
-	list_for_each_entry(rtd, &card->rtd_list, list)
+	/*
+	 * Flush out pmdown_time work - we actually do want to run it
+	 * now, we're shutting down so no imminent restart.
+	 */
+	for_each_card_rtds(card, rtd)
 		flush_delayed_work(&rtd->delayed_work);
 
 	snd_soc_dapm_shutdown(card);
 
 	/* deactivate pins to sleep state */
-	list_for_each_entry(rtd, &card->rtd_list, list) {
+	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+		struct snd_soc_dai *codec_dai;
 		int i;
 
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
-		for (i = 0; i < rtd->num_codecs; i++) {
-			struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			pinctrl_pm_select_sleep_state(codec_dai->dev);
 		}
 	}
@@ -2315,6 +2322,7 @@ static int snd_soc_add_controls(struct snd_card *card, struct device *dev,
 
 	for (i = 0; i < num_controls; i++) {
 		const struct snd_kcontrol_new *control = &controls[i];
+
 		err = snd_ctl_add(card, snd_soc_cnew(control, data,
 						     control->name, prefix));
 		if (err < 0) {
@@ -2432,8 +2440,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
  *
  * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
  */
-int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id,
-			     int source, unsigned int freq, int dir)
+int snd_soc_component_set_sysclk(struct snd_soc_component *component,
+				 int clk_id, int source, unsigned int freq,
+				 int dir)
 {
 	if (component->driver->set_sysclk)
 		return component->driver->set_sysclk(component, clk_id, source,
@@ -2501,7 +2510,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 {
 	if (component->driver->set_pll)
 		return component->driver->set_pll(component, pll_id, source,
-					      freq_in, freq_out);
+						  freq_in, freq_out);
 
 	return -EINVAL;
 }
@@ -2532,8 +2541,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->driver == NULL)
-		return -EINVAL;
 	if (dai->driver->ops->set_fmt == NULL)
 		return -ENOTSUPP;
 	return dai->driver->ops->set_fmt(dai, fmt);
@@ -2549,8 +2556,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
  * Generates the TDM tx and rx slot default masks for DAI.
  */
 static int snd_soc_xlate_tdm_slot_mask(unsigned int slots,
-					  unsigned int *tx_mask,
-					  unsigned int *rx_mask)
+				       unsigned int *tx_mask,
+				       unsigned int *rx_mask)
 {
 	if (*tx_mask || *rx_mask)
 		return 0;
@@ -2680,9 +2687,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 			     int direction)
 {
-	if (!dai->driver)
-		return -ENOTSUPP;
-
 	if (dai->driver->ops->mute_stream)
 		return dai->driver->ops->mute_stream(dai, mute, direction);
 	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
@@ -2693,6 +2697,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
+static int snd_soc_bind_card(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *rtd;
+	int ret;
+
+	ret = snd_soc_instantiate_card(card);
+	if (ret != 0)
+		return ret;
+
+	/* deactivate pins to sleep state */
+	for_each_card_rtds(card, rtd) {
+		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+		struct snd_soc_dai *codec_dai;
+		int j;
+
+		for_each_rtd_codec_dai(rtd, j, codec_dai) {
+			if (!codec_dai->active)
+				pinctrl_pm_select_sleep_state(codec_dai->dev);
+		}
+
+		if (!cpu_dai->active)
+			pinctrl_pm_select_sleep_state(cpu_dai->dev);
+	}
+
+	return ret;
+}
+
 /**
  * snd_soc_register_card - Register a card with the ASoC core
  *
@@ -2702,13 +2733,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 int snd_soc_register_card(struct snd_soc_card *card)
 {
 	int i, ret;
-	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai_link *link;
 
 	if (!card->name || !card->dev)
 		return -EINVAL;
 
-	for (i = 0; i < card->num_links; i++) {
-		struct snd_soc_dai_link *link = &card->dai_link[i];
+	for_each_card_prelinks(card, i, link) {
 
 		ret = soc_init_dai_link(card, link);
 		if (ret) {
@@ -2723,7 +2753,6 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	snd_soc_initialize_card_lists(card);
 
 	INIT_LIST_HEAD(&card->dai_link_list);
-	card->num_dai_links = 0;
 
 	INIT_LIST_HEAD(&card->rtd_list);
 	card->num_rtd = 0;
@@ -2734,28 +2763,23 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	mutex_init(&card->mutex);
 	mutex_init(&card->dapm_mutex);
 
-	ret = snd_soc_instantiate_card(card);
-	if (ret != 0)
-		return ret;
-
-	/* deactivate pins to sleep state */
-	list_for_each_entry(rtd, &card->rtd_list, list)  {
-		struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-		int j;
-
-		for (j = 0; j < rtd->num_codecs; j++) {
-			struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
-			if (!codec_dai->active)
-				pinctrl_pm_select_sleep_state(codec_dai->dev);
-		}
+	return snd_soc_bind_card(card);
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
-		if (!cpu_dai->active)
-			pinctrl_pm_select_sleep_state(cpu_dai->dev);
+static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister)
+{
+	if (card->instantiated) {
+		card->instantiated = false;
+		snd_soc_dapm_shutdown(card);
+		soc_cleanup_card_resources(card);
+		if (!unregister)
+			list_add(&card->list, &unbind_card_list);
+	} else {
+		if (unregister)
+			list_del(&card->list);
 	}
-
-	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
 
 /**
  * snd_soc_unregister_card - Unregister a card with the ASoC core
@@ -2765,12 +2789,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
  */
 int snd_soc_unregister_card(struct snd_soc_card *card)
 {
-	if (card->instantiated) {
-		card->instantiated = false;
-		snd_soc_dapm_shutdown(card);
-		soc_cleanup_card_resources(card);
-		dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
-	}
+	snd_soc_unbind_card(card, true);
+	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
 	return 0;
 }
@@ -2802,7 +2822,7 @@ static char *fmt_single_name(struct device *dev, int *id)
 		}
 
 	} else {
-		/* I2C component devices are named "bus-addr"  */
+		/* I2C component devices are named "bus-addr" */
 		if (sscanf(name, "%x-%x", &id1, &id2) == 2) {
 			char tmp[NAME_SIZE];
 
@@ -2810,7 +2830,8 @@ static char *fmt_single_name(struct device *dev, int *id)
 			*id = ((id1 & 0xffff) << 16) + id2;
 
 			/* sanitize component name for DAI link creation */
-			snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name);
+			snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name,
+				 name);
 			strlcpy(name, tmp, NAME_SIZE);
 		} else
 			*id = 0;
@@ -2845,7 +2866,7 @@ static void snd_soc_unregister_dais(struct snd_soc_component *component)
 {
 	struct snd_soc_dai *dai, *_dai;
 
-	list_for_each_entry_safe(dai, _dai, &component->dai_list, list) {
+	for_each_component_dais_safe(component, dai, _dai) {
 		dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n",
 			dai->name);
 		list_del(&dai->list);
@@ -2877,7 +2898,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
 	 * component-less anymore.
 	 */
 	if (legacy_dai_naming &&
-	   (dai_drv->id == 0 || dai_drv->name == NULL)) {
+	    (dai_drv->id == 0 || dai_drv->name == NULL)) {
 		dai->name = fmt_single_name(dev, &dai->id);
 	} else {
 		dai->name = fmt_multiple_name(dev, dai_drv);
@@ -2897,6 +2918,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
 	if (!dai->driver->ops)
 		dai->driver->ops = &null_dai_ops;
 
+	/* see for_each_component_dais */
 	list_add_tail(&dai->list, &component->dai_list);
 	component->num_dai++;
 
@@ -2910,11 +2932,10 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component,
  * @component: The component the DAIs are registered for
  * @dai_drv: DAI driver to use for the DAIs
  * @count: Number of DAIs
- * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the
- *                     parent's name.
  */
 static int snd_soc_register_dais(struct snd_soc_component *component,
-				 struct snd_soc_dai_driver *dai_drv, size_t count)
+				 struct snd_soc_dai_driver *dai_drv,
+				 size_t count)
 {
 	struct device *dev = component->dev;
 	struct snd_soc_dai *dai;
@@ -2925,8 +2946,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component,
 
 	for (i = 0; i < count; i++) {
 
-		dai = soc_add_dai(component, dai_drv + i,
-				  count == 1 && !component->driver->non_legacy_dai_naming);
+		dai = soc_add_dai(component, dai_drv + i, count == 1 &&
+				  !component->driver->non_legacy_dai_naming);
 		if (dai == NULL) {
 			ret = -ENOMEM;
 			goto err;
@@ -2970,7 +2991,8 @@ int snd_soc_register_dai(struct snd_soc_component *component,
 	if (!dai)
 		return -ENOMEM;
 
-	/* Create the DAI widgets here. After adding DAIs, topology may
+	/*
+	 * Create the DAI widgets here. After adding DAIs, topology may
 	 * also add routes that need these widgets as source or sink.
 	 */
 	ret = snd_soc_dapm_new_dai_widgets(dapm, dai);
@@ -3052,7 +3074,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component)
 #ifdef CONFIG_REGMAP
 
 /**
- * snd_soc_component_init_regmap() - Initialize regmap instance for the component
+ * snd_soc_component_init_regmap() - Initialize regmap instance for the
+ *                                   component
  * @component: The component for which to initialize the regmap instance
  * @regmap: The regmap instance that should be used by the component
  *
@@ -3070,7 +3093,8 @@ void snd_soc_component_init_regmap(struct snd_soc_component *component,
 EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap);
 
 /**
- * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component
+ * snd_soc_component_exit_regmap() - De-initialize regmap instance for the
+ *                                   component
  * @component: The component for which to de-initialize the regmap instance
  *
  * Calls regmap_exit() on the regmap instance associated to the component and
@@ -3094,11 +3118,13 @@ static void snd_soc_component_add(struct snd_soc_component *component)
 
 	if (!component->driver->write && !component->driver->read) {
 		if (!component->regmap)
-			component->regmap = dev_get_regmap(component->dev, NULL);
+			component->regmap = dev_get_regmap(component->dev,
+							   NULL);
 		if (component->regmap)
 			snd_soc_component_setup_regmap(component);
 	}
 
+	/* see for_each_component */
 	list_add(&component->list, &component_list);
 	INIT_LIST_HEAD(&component->dobj_list);
 
@@ -3116,7 +3142,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
 	struct snd_soc_card *card = component->card;
 
 	if (card)
-		snd_soc_unregister_card(card);
+		snd_soc_unbind_card(card, false);
 
 	list_del(&component->list);
 }
@@ -3156,6 +3182,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream)
 			stream->formats |= endianness_format_map[i];
 }
 
+static void snd_soc_try_rebind_card(void)
+{
+	struct snd_soc_card *card, *c;
+
+	if (!list_empty(&unbind_card_list)) {
+		list_for_each_entry_safe(card, c, &unbind_card_list, list) {
+			if (!snd_soc_bind_card(card))
+				list_del(&card->list);
+		}
+	}
+}
+
 int snd_soc_add_component(struct device *dev,
 			struct snd_soc_component *component,
 			const struct snd_soc_component_driver *component_driver,
@@ -3183,6 +3221,7 @@ int snd_soc_add_component(struct device *dev,
 	}
 
 	snd_soc_component_add(component);
+	snd_soc_try_rebind_card();
 
 	return 0;
 
@@ -3221,27 +3260,28 @@ static int __snd_soc_unregister_component(struct device *dev)
 	int found = 0;
 
 	mutex_lock(&client_mutex);
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 		if (dev != component->dev)
 			continue;
 
-		snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL);
+		snd_soc_tplg_component_remove(component,
+					      SND_SOC_TPLG_INDEX_ALL);
 		snd_soc_component_del_unlocked(component);
 		found = 1;
 		break;
 	}
 	mutex_unlock(&client_mutex);
 
-	if (found) {
+	if (found)
 		snd_soc_component_cleanup(component);
-	}
 
 	return found;
 }
 
 void snd_soc_unregister_component(struct device *dev)
 {
-	while (__snd_soc_unregister_component(dev));
+	while (__snd_soc_unregister_component(dev))
+		;
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_component);
 
@@ -3253,7 +3293,7 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 
 	ret = NULL;
 	mutex_lock(&client_mutex);
-	list_for_each_entry(component, &component_list, list) {
+	for_each_component(component) {
 		if (dev != component->dev)
 			continue;
 
@@ -3653,7 +3693,7 @@ int snd_soc_get_dai_id(struct device_node *ep)
 	 */
 	ret = -ENOTSUPP;
 	mutex_lock(&client_mutex);
-	list_for_each_entry(pos, &component_list, list) {
+	for_each_component(pos) {
 		struct device_node *component_of_node = pos->dev->of_node;
 
 		if (!component_of_node && pos->dev->parent)
@@ -3683,7 +3723,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
 	int ret = -EPROBE_DEFER;
 
 	mutex_lock(&client_mutex);
-	list_for_each_entry(pos, &component_list, list) {
+	for_each_component(pos) {
 		component_of_node = pos->dev->of_node;
 		if (!component_of_node && pos->dev->parent)
 			component_of_node = pos->dev->parent->of_node;
@@ -3719,7 +3759,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
 			ret = 0;
 
 			/* find target DAI */
-			list_for_each_entry(dai, &pos->dai_list, list) {
+			for_each_component_dais(pos, dai) {
 				if (id == 0)
 					break;
 				id--;
@@ -3764,10 +3804,10 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name);
  */
 void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link)
 {
-	struct snd_soc_dai_link_component *component = dai_link->codecs;
+	struct snd_soc_dai_link_component *component;
 	int index;
 
-	for (index = 0; index < dai_link->num_codecs; index++, component++) {
+	for_each_link_codecs(dai_link, index, component) {
 		if (!component->of_node)
 			break;
 		of_node_put(component->of_node);
@@ -3819,12 +3859,10 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev,
 	dai_link->num_codecs = num_codecs;
 
 	/* Parse the list */
-	for (index = 0, component = dai_link->codecs;
-	     index < dai_link->num_codecs;
-	     index++, component++) {
+	for_each_link_codecs(dai_link, index, component) {
 		ret = of_parse_phandle_with_args(of_node, name,
 						 "#sound-dai-cells",
-						  index, &args);
+						 index, &args);
 		if (ret)
 			goto err;
 		component->of_node = args.np;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 461d951917c0..a5178845065b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -18,7 +18,6 @@
 //      device reopen.
 
 #include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/async.h>
 #include <linux/delay.h>
@@ -364,10 +363,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 				ret = PTR_ERR(data->widget);
 				goto err_data;
 			}
-			if (!data->widget) {
-				ret = -ENOMEM;
-				goto err_data;
-			}
 		}
 		break;
 	case snd_soc_dapm_demux:
@@ -402,10 +397,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 				ret = PTR_ERR(data->widget);
 				goto err_data;
 			}
-			if (!data->widget) {
-				ret = -ENOMEM;
-				goto err_data;
-			}
 
 			snd_soc_dapm_add_path(widget->dapm, data->widget,
 					      widget, NULL, NULL);
@@ -1026,9 +1017,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w)
 	struct snd_kcontrol *kcontrol;
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_card *card = dapm->card->snd_card;
+	struct snd_soc_pcm_runtime *rtd = w->priv;
 
 	/* create control for links with > 1 config */
-	if (w->num_params <= 1)
+	if (rtd->dai_link->num_params <= 1)
 		return 0;
 
 	/* add kcontrol */
@@ -1320,14 +1312,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w,
 
 	soc_dapm_async_complete(w->dapm);
 
-#ifdef CONFIG_HAVE_CLK
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		return clk_prepare_enable(w->clk);
 	} else {
 		clk_disable_unprepare(w->clk);
 		return 0;
 	}
-#endif
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(dapm_clock_event);
@@ -1953,7 +1944,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 	dapm_pre_sequence_async(&card->dapm, 0);
 	/* Run other bias changes in parallel */
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d != &card->dapm)
+		if (d != &card->dapm && d->bias_level != d->target_bias_level)
 			async_schedule_domain(dapm_pre_sequence_async, d,
 						&async_domain);
 	}
@@ -1977,7 +1968,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 
 	/* Run all the bias changes in parallel */
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d != &card->dapm)
+		if (d != &card->dapm && d->bias_level != d->target_bias_level)
 			async_schedule_domain(dapm_post_sequence_async, d,
 						&async_domain);
 	}
@@ -2371,12 +2362,13 @@ static ssize_t dapm_widget_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
+	struct snd_soc_dai *codec_dai;
 	int i, count = 0;
 
 	mutex_lock(&rtd->card->dapm_mutex);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component;
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		struct snd_soc_component *cmpnt = codec_dai->component;
 
 		count += dapm_widget_show_component(cmpnt, buf + count);
 	}
@@ -3426,35 +3418,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
 
 struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
-	const struct snd_soc_dapm_widget *widget)
-{
-	struct snd_soc_dapm_widget *w;
-
-	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-	w = snd_soc_dapm_new_control_unlocked(dapm, widget);
-	/* Do not nag about probe deferrals */
-	if (IS_ERR(w)) {
-		int ret = PTR_ERR(w);
-
-		if (ret != -EPROBE_DEFER)
-			dev_err(dapm->dev,
-				"ASoC: Failed to create DAPM control %s (%d)\n",
-				widget->name, ret);
-		goto out_unlock;
-	}
-	if (!w)
-		dev_err(dapm->dev,
-			"ASoC: Failed to create DAPM control %s\n",
-			widget->name);
-
-out_unlock:
-	mutex_unlock(&dapm->card->dapm_mutex);
-	return w;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
-
-struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 			 const struct snd_soc_dapm_widget *widget)
 {
@@ -3464,53 +3427,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 	int ret;
 
 	if ((w = dapm_cnew_widget(widget)) == NULL)
-		return NULL;
+		return ERR_PTR(-ENOMEM);
 
 	switch (w->id) {
 	case snd_soc_dapm_regulator_supply:
 		w->regulator = devm_regulator_get(dapm->dev, w->name);
 		if (IS_ERR(w->regulator)) {
 			ret = PTR_ERR(w->regulator);
-			if (ret == -EPROBE_DEFER)
-				return ERR_PTR(ret);
-			dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-				w->name, ret);
-			return NULL;
+			goto request_failed;
 		}
 
 		if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) {
 			ret = regulator_allow_bypass(w->regulator, true);
 			if (ret != 0)
-				dev_warn(w->dapm->dev,
+				dev_warn(dapm->dev,
 					 "ASoC: Failed to bypass %s: %d\n",
 					 w->name, ret);
 		}
 		break;
 	case snd_soc_dapm_pinctrl:
 		w->pinctrl = devm_pinctrl_get(dapm->dev);
-		if (IS_ERR_OR_NULL(w->pinctrl)) {
+		if (IS_ERR(w->pinctrl)) {
 			ret = PTR_ERR(w->pinctrl);
-			if (ret == -EPROBE_DEFER)
-				return ERR_PTR(ret);
-			dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-				w->name, ret);
-			return NULL;
+			goto request_failed;
 		}
 		break;
 	case snd_soc_dapm_clock_supply:
-#ifdef CONFIG_CLKDEV_LOOKUP
 		w->clk = devm_clk_get(dapm->dev, w->name);
 		if (IS_ERR(w->clk)) {
 			ret = PTR_ERR(w->clk);
-			if (ret == -EPROBE_DEFER)
-				return ERR_PTR(ret);
-			dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
-				w->name, ret);
-			return NULL;
+			goto request_failed;
 		}
-#else
-		return NULL;
-#endif
 		break;
 	default:
 		break;
@@ -3523,7 +3470,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 		w->name = kstrdup_const(widget->name, GFP_KERNEL);
 	if (w->name == NULL) {
 		kfree(w);
-		return NULL;
+		return ERR_PTR(-ENOMEM);
 	}
 
 	switch (w->id) {
@@ -3600,7 +3547,37 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 	/* machine layer sets up unconnected pins and insertions */
 	w->connected = 1;
 	return w;
+
+request_failed:
+	if (ret != -EPROBE_DEFER)
+		dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
+			w->name, ret);
+
+	return ERR_PTR(ret);
+}
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @dapm: DAPM context
+ * @widget: widget template
+ *
+ * Creates new DAPM control based upon a template.
+ *
+ * Returns a widget pointer on success or an error pointer on failure
+ */
+struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+			 const struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_dapm_widget *w;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	w = snd_soc_dapm_new_control_unlocked(dapm, widget);
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return w;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
 
 /**
  * snd_soc_dapm_new_controls - create new dapm controls
@@ -3625,19 +3602,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 		w = snd_soc_dapm_new_control_unlocked(dapm, widget);
 		if (IS_ERR(w)) {
 			ret = PTR_ERR(w);
-			/* Do not nag about probe deferrals */
-			if (ret == -EPROBE_DEFER)
-				break;
-			dev_err(dapm->dev,
-				"ASoC: Failed to create DAPM control %s (%d)\n",
-				widget->name, ret);
-			break;
-		}
-		if (!w) {
-			dev_err(dapm->dev,
-				"ASoC: Failed to create DAPM control %s\n",
-				widget->name);
-			ret = -ENOMEM;
 			break;
 		}
 		widget++;
@@ -3650,32 +3614,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
 static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 				  struct snd_kcontrol *kcontrol, int event)
 {
-	struct snd_soc_dapm_path *source_p, *sink_p;
+	struct snd_soc_dapm_path *path;
 	struct snd_soc_dai *source, *sink;
 	struct snd_soc_pcm_runtime *rtd = w->priv;
-	const struct snd_soc_pcm_stream *config = w->params + w->params_select;
+	const struct snd_soc_pcm_stream *config;
 	struct snd_pcm_substream substream;
 	struct snd_pcm_hw_params *params = NULL;
 	struct snd_pcm_runtime *runtime = NULL;
 	unsigned int fmt;
-	int ret;
+	int ret = 0;
+
+	config = rtd->dai_link->params + rtd->params_select;
 
 	if (WARN_ON(!config) ||
 	    WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) ||
 		    list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])))
 		return -EINVAL;
 
-	/* We only support a single source and sink, pick the first */
-	source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT],
-				    struct snd_soc_dapm_path,
-				    list_node[SND_SOC_DAPM_DIR_OUT]);
-	sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN],
-				    struct snd_soc_dapm_path,
-				    list_node[SND_SOC_DAPM_DIR_IN]);
-
-	source = source_p->source->priv;
-	sink = sink_p->sink->priv;
-
 	/* Be a little careful as we don't want to overflow the mask array */
 	if (config->formats) {
 		fmt = ffs(config->formats) - 1;
@@ -3717,59 +3672,95 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-		if (source->driver->ops->startup) {
-			ret = source->driver->ops->startup(&substream, source);
-			if (ret < 0) {
-				dev_err(source->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			if (source->driver->ops->startup) {
+				ret = source->driver->ops->startup(&substream,
+								   source);
+				if (ret < 0) {
+					dev_err(source->dev,
+						"ASoC: startup() failed: %d\n",
+						ret);
+					goto out;
+				}
+				source->active++;
 			}
-			source->active++;
+			ret = soc_dai_hw_params(&substream, params, source);
+			if (ret < 0)
+				goto out;
 		}
-		ret = soc_dai_hw_params(&substream, params, source);
-		if (ret < 0)
-			goto out;
 
 		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-		if (sink->driver->ops->startup) {
-			ret = sink->driver->ops->startup(&substream, sink);
-			if (ret < 0) {
-				dev_err(sink->dev,
-					"ASoC: startup() failed: %d\n", ret);
-				goto out;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			if (sink->driver->ops->startup) {
+				ret = sink->driver->ops->startup(&substream,
+								 sink);
+				if (ret < 0) {
+					dev_err(sink->dev,
+						"ASoC: startup() failed: %d\n",
+						ret);
+					goto out;
+				}
+				sink->active++;
 			}
-			sink->active++;
+			ret = soc_dai_hw_params(&substream, params, sink);
+			if (ret < 0)
+				goto out;
 		}
-		ret = soc_dai_hw_params(&substream, params, sink);
-		if (ret < 0)
-			goto out;
 		break;
 
 	case SND_SOC_DAPM_POST_PMU:
-		ret = snd_soc_dai_digital_mute(sink, 0,
-					       SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret != 0 && ret != -ENOTSUPP)
-			dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret);
-		ret = 0;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			ret = snd_soc_dai_digital_mute(sink, 0,
+						       SNDRV_PCM_STREAM_PLAYBACK);
+			if (ret != 0 && ret != -ENOTSUPP)
+				dev_warn(sink->dev,
+					 "ASoC: Failed to unmute: %d\n", ret);
+			ret = 0;
+		}
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
-		ret = snd_soc_dai_digital_mute(sink, 1,
-					       SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret != 0 && ret != -ENOTSUPP)
-			dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret);
-		ret = 0;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			ret = snd_soc_dai_digital_mute(sink, 1,
+						       SNDRV_PCM_STREAM_PLAYBACK);
+			if (ret != 0 && ret != -ENOTSUPP)
+				dev_warn(sink->dev,
+					 "ASoC: Failed to mute: %d\n", ret);
+			ret = 0;
+		}
 
-		source->active--;
-		if (source->driver->ops->shutdown) {
-			substream.stream = SNDRV_PCM_STREAM_CAPTURE;
-			source->driver->ops->shutdown(&substream, source);
+		substream.stream = SNDRV_PCM_STREAM_CAPTURE;
+		snd_soc_dapm_widget_for_each_source_path(w, path) {
+			source = path->source->priv;
+
+			if (source->driver->ops->hw_free)
+				source->driver->ops->hw_free(&substream,
+							     source);
+
+			source->active--;
+			if (source->driver->ops->shutdown)
+				source->driver->ops->shutdown(&substream,
+							      source);
 		}
 
-		sink->active--;
-		if (sink->driver->ops->shutdown) {
-			substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
-			sink->driver->ops->shutdown(&substream, sink);
+		substream.stream = SNDRV_PCM_STREAM_PLAYBACK;
+		snd_soc_dapm_widget_for_each_sink_path(w, path) {
+			sink = path->sink->priv;
+
+			if (sink->driver->ops->hw_free)
+				sink->driver->ops->hw_free(&substream, sink);
+
+			sink->active--;
+			if (sink->driver->ops->shutdown)
+				sink->driver->ops->shutdown(&substream, sink);
 		}
 		break;
 
@@ -3788,8 +3779,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_pcm_runtime *rtd = w->priv;
 
-	ucontrol->value.enumerated.item[0] = w->params_select;
+	ucontrol->value.enumerated.item[0] = rtd->params_select;
 
 	return 0;
 }
@@ -3798,18 +3790,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_pcm_runtime *rtd = w->priv;
 
 	/* Can't change the config when widget is already powered */
 	if (w->power)
 		return -EBUSY;
 
-	if (ucontrol->value.enumerated.item[0] == w->params_select)
+	if (ucontrol->value.enumerated.item[0] == rtd->params_select)
 		return 0;
 
-	if (ucontrol->value.enumerated.item[0] >= w->num_params)
+	if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params)
 		return -EINVAL;
 
-	w->params_select = ucontrol->value.enumerated.item[0];
+	rtd->params_select = ucontrol->value.enumerated.item[0];
 
 	return 0;
 }
@@ -3896,12 +3889,10 @@ outfree_w_param:
 	return NULL;
 }
 
-int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
-			 struct snd_soc_pcm_runtime *rtd,
-			 const struct snd_soc_pcm_stream *params,
-			 unsigned int num_params,
-			 struct snd_soc_dapm_widget *source,
-			 struct snd_soc_dapm_widget *sink)
+static struct snd_soc_dapm_widget *
+snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd,
+		     struct snd_soc_dapm_widget *source,
+		     struct snd_soc_dapm_widget *sink)
 {
 	struct snd_soc_dapm_widget template;
 	struct snd_soc_dapm_widget *w;
@@ -3913,7 +3904,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 	link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s",
 				   source->name, sink->name);
 	if (!link_name)
-		return -ENOMEM;
+		return ERR_PTR(-ENOMEM);
 
 	memset(&template, 0, sizeof(template));
 	template.reg = SND_SOC_NOPM;
@@ -3925,9 +3916,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 	template.kcontrol_news = NULL;
 
 	/* allocate memory for control, only in case of multiple configs */
-	if (num_params > 1) {
-		w_param_text = devm_kcalloc(card->dev, num_params,
-					sizeof(char *), GFP_KERNEL);
+	if (rtd->dai_link->num_params > 1) {
+		w_param_text = devm_kcalloc(card->dev,
+					    rtd->dai_link->num_params,
+					    sizeof(char *), GFP_KERNEL);
 		if (!w_param_text) {
 			ret = -ENOMEM;
 			goto param_fail;
@@ -3936,7 +3928,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 		template.num_kcontrols = 1;
 		template.kcontrol_news =
 					snd_soc_dapm_alloc_kcontrol(card,
-						link_name, params, num_params,
+						link_name,
+						rtd->dai_link->params,
+						rtd->dai_link->num_params,
 						w_param_text, &private_value);
 		if (!template.kcontrol_news) {
 			ret = -ENOMEM;
@@ -3950,37 +3944,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 	w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
 	if (IS_ERR(w)) {
 		ret = PTR_ERR(w);
-		/* Do not nag about probe deferrals */
-		if (ret != -EPROBE_DEFER)
-			dev_err(card->dev,
-				"ASoC: Failed to create %s widget (%d)\n",
-				link_name, ret);
-		goto outfree_kcontrol_news;
-	}
-	if (!w) {
-		dev_err(card->dev, "ASoC: Failed to create %s widget\n",
-			link_name);
-		ret = -ENOMEM;
 		goto outfree_kcontrol_news;
 	}
 
-	w->params = params;
-	w->num_params = num_params;
 	w->priv = rtd;
 
-	ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL);
-	if (ret)
-		goto outfree_w;
-	return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL);
+	return w;
 
-outfree_w:
-	devm_kfree(card->dev, w);
 outfree_kcontrol_news:
 	devm_kfree(card->dev, (void *)template.kcontrol_news);
-	snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text);
+	snd_soc_dapm_free_kcontrol(card, &private_value,
+				   rtd->dai_link->num_params, w_param_text);
 param_fail:
 	devm_kfree(card->dev, link_name);
-	return ret;
+	return ERR_PTR(ret);
 }
 
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
@@ -4003,21 +3980,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 			template.name);
 
 		w = snd_soc_dapm_new_control_unlocked(dapm, &template);
-		if (IS_ERR(w)) {
-			int ret = PTR_ERR(w);
-
-			/* Do not nag about probe deferrals */
-			if (ret != -EPROBE_DEFER)
-				dev_err(dapm->dev,
-				"ASoC: Failed to create %s widget (%d)\n",
-				dai->driver->playback.stream_name, ret);
-			return ret;
-		}
-		if (!w) {
-			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
-				dai->driver->playback.stream_name);
-			return -ENOMEM;
-		}
+		if (IS_ERR(w))
+			return PTR_ERR(w);
 
 		w->priv = dai;
 		dai->playback_widget = w;
@@ -4032,21 +3996,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 			template.name);
 
 		w = snd_soc_dapm_new_control_unlocked(dapm, &template);
-		if (IS_ERR(w)) {
-			int ret = PTR_ERR(w);
-
-			/* Do not nag about probe deferrals */
-			if (ret != -EPROBE_DEFER)
-				dev_err(dapm->dev,
-				"ASoC: Failed to create %s widget (%d)\n",
-				dai->driver->playback.stream_name, ret);
-			return ret;
-		}
-		if (!w) {
-			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
-				dai->driver->capture.stream_name);
-			return -ENOMEM;
-		}
+		if (IS_ERR(w))
+			return PTR_ERR(w);
 
 		w->priv = dai;
 		dai->capture_widget = w;
@@ -4115,34 +4066,79 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
 					  struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dapm_widget *sink, *source;
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
+	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
 	int i;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+	if (rtd->dai_link->params) {
+		playback_cpu = cpu_dai->capture_widget;
+		capture_cpu = cpu_dai->playback_widget;
+	} else {
+		playback = cpu_dai->playback_widget;
+		capture = cpu_dai->capture_widget;
+		playback_cpu = playback;
+		capture_cpu = capture;
+	}
+
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 
 		/* connect BE DAI playback if widgets are valid */
-		if (codec_dai->playback_widget && cpu_dai->playback_widget) {
-			source = cpu_dai->playback_widget;
-			sink = codec_dai->playback_widget;
+		codec = codec_dai->playback_widget;
+
+		if (playback_cpu && codec) {
+			if (!playback) {
+				playback = snd_soc_dapm_new_dai(card, rtd,
+								playback_cpu,
+								codec);
+				if (IS_ERR(playback)) {
+					dev_err(rtd->dev,
+						"ASoC: Failed to create DAI %s: %ld\n",
+						codec_dai->name,
+						PTR_ERR(playback));
+					continue;
+				}
+
+				snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+						      playback, NULL, NULL);
+			}
+
 			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-				cpu_dai->component->name, source->name,
-				codec_dai->component->name, sink->name);
+				cpu_dai->component->name, playback_cpu->name,
+				codec_dai->component->name, codec->name);
 
-			snd_soc_dapm_add_path(&card->dapm, source, sink,
-				NULL, NULL);
+			snd_soc_dapm_add_path(&card->dapm, playback, codec,
+					      NULL, NULL);
 		}
+	}
 
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		/* connect BE DAI capture if widgets are valid */
-		if (codec_dai->capture_widget && cpu_dai->capture_widget) {
-			source = codec_dai->capture_widget;
-			sink = cpu_dai->capture_widget;
+		codec = codec_dai->capture_widget;
+
+		if (codec && capture_cpu) {
+			if (!capture) {
+				capture = snd_soc_dapm_new_dai(card, rtd,
+							       codec,
+							       capture_cpu);
+				if (IS_ERR(capture)) {
+					dev_err(rtd->dev,
+						"ASoC: Failed to create DAI %s: %ld\n",
+						codec_dai->name,
+						PTR_ERR(capture));
+					continue;
+				}
+
+				snd_soc_dapm_add_path(&card->dapm, capture,
+						      capture_cpu, NULL, NULL);
+			}
+
 			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-				codec_dai->component->name, source->name,
-				cpu_dai->component->name, sink->name);
+				codec_dai->component->name, codec->name,
+				cpu_dai->component->name, capture_cpu->name);
 
-			snd_soc_dapm_add_path(&card->dapm, source, sink,
-				NULL, NULL);
+			snd_soc_dapm_add_path(&card->dapm, codec, capture,
+					      NULL, NULL);
 		}
 	}
 }
@@ -4192,12 +4188,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 	struct snd_soc_pcm_runtime *rtd;
 
 	/* for each BE DAI link... */
-	list_for_each_entry(rtd, &card->rtd_list, list)  {
+	for_each_card_rtds(card, rtd)  {
 		/*
 		 * dynamic FE links have no fixed DAI mapping.
 		 * CODEC<->CODEC links have no direct connection.
 		 */
-		if (rtd->dai_link->dynamic || rtd->dai_link->params)
+		if (rtd->dai_link->dynamic)
 			continue;
 
 		dapm_connect_dai_link_widgets(card, rtd);
@@ -4207,11 +4203,12 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 	int event)
 {
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
-	for (i = 0; i < rtd->num_codecs; i++)
-		soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event);
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		soc_dapm_dai_stream_event(codec_dai, stream, event);
 
 	dapm_power_widgets(rtd->card, event);
 }
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 592efb370c44..f4dc3d445aae 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
 	unsigned int rshift = mc->rshift;
 	int max = mc->max;
 	int min = mc->min;
-	unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+	unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
 	unsigned int val;
 	int ret;
 
@@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
 	unsigned int rshift = mc->rshift;
 	int max = mc->max;
 	int min = mc->min;
-	unsigned int mask = (1 << (fls(min + max) - 1)) - 1;
+	unsigned int mask = (1U << (fls(min + max) - 1)) - 1;
 	int err = 0;
 	unsigned int val, val_mask, val2 = 0;
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e8b98bfd4cf1..03f36e534050 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -59,25 +59,26 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream)
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	lockdep_assert_held(&rtd->pcm_mutex);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active++;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->playback_active++;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->playback_active++;
 	} else {
 		cpu_dai->capture_active++;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->capture_active++;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->capture_active++;
 	}
 
 	cpu_dai->active++;
 	cpu_dai->component->active++;
-	for (i = 0; i < rtd->num_codecs; i++) {
-		rtd->codec_dais[i]->active++;
-		rtd->codec_dais[i]->component->active++;
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		codec_dai->active++;
+		codec_dai->component->active++;
 	}
 }
 
@@ -94,25 +95,26 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	lockdep_assert_held(&rtd->pcm_mutex);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		cpu_dai->playback_active--;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->playback_active--;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->playback_active--;
 	} else {
 		cpu_dai->capture_active--;
-		for (i = 0; i < rtd->num_codecs; i++)
-			rtd->codec_dais[i]->capture_active--;
+		for_each_rtd_codec_dai(rtd, i, codec_dai)
+			codec_dai->capture_active--;
 	}
 
 	cpu_dai->active--;
 	cpu_dai->component->active--;
-	for (i = 0; i < rtd->num_codecs; i++) {
-		rtd->codec_dais[i]->component->active--;
-		rtd->codec_dais[i]->active--;
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		codec_dai->component->active--;
+		codec_dai->active--;
 	}
 }
 
@@ -172,7 +174,7 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 {
 	struct snd_soc_dpcm *dpcm;
 
-	list_for_each_entry(dpcm, &fe->dpcm[dir].be_clients, list_be) {
+	for_each_dpcm_be(fe, dir, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 
@@ -253,6 +255,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	unsigned int rate, channels, sample_bits, symmetry, i;
 
 	rate = params_rate(params);
@@ -263,8 +266,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	symmetry = cpu_dai->driver->symmetric_rates ||
 		rtd->dai_link->symmetric_rates;
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		symmetry |= rtd->codec_dais[i]->driver->symmetric_rates;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		symmetry |= codec_dai->driver->symmetric_rates;
 
 	if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
 		dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
@@ -275,8 +278,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	symmetry = cpu_dai->driver->symmetric_channels ||
 		rtd->dai_link->symmetric_channels;
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		symmetry |= rtd->codec_dais[i]->driver->symmetric_channels;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		symmetry |= codec_dai->driver->symmetric_channels;
 
 	if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
 		dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
@@ -287,8 +290,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	symmetry = cpu_dai->driver->symmetric_samplebits ||
 		rtd->dai_link->symmetric_samplebits;
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		symmetry |= rtd->codec_dais[i]->driver->symmetric_samplebits;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		symmetry |= codec_dai->driver->symmetric_samplebits;
 
 	if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
 		dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
@@ -304,17 +307,18 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
 	struct snd_soc_dai_link *link = rtd->dai_link;
+	struct snd_soc_dai *codec_dai;
 	unsigned int symmetry, i;
 
 	symmetry = cpu_driver->symmetric_rates || link->symmetric_rates ||
 		cpu_driver->symmetric_channels || link->symmetric_channels ||
 		cpu_driver->symmetric_samplebits || link->symmetric_samplebits;
 
-	for (i = 0; i < rtd->num_codecs; i++)
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
 		symmetry = symmetry ||
-			rtd->codec_dais[i]->driver->symmetric_rates ||
-			rtd->codec_dais[i]->driver->symmetric_channels ||
-			rtd->codec_dais[i]->driver->symmetric_samplebits;
+			codec_dai->driver->symmetric_rates ||
+			codec_dai->driver->symmetric_channels ||
+			codec_dai->driver->symmetric_samplebits;
 
 	return symmetry;
 }
@@ -342,8 +346,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 	unsigned int bits = 0, cpu_bits;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (codec_dai->driver->playback.sig_bits == 0) {
 				bits = 0;
 				break;
@@ -352,8 +355,7 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 		}
 		cpu_bits = cpu_dai->driver->playback.sig_bits;
 	} else {
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (codec_dai->driver->capture.sig_bits == 0) {
 				bits = 0;
 				break;
@@ -372,6 +374,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_pcm_hardware *hw = &runtime->hw;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver;
 	struct snd_soc_dai_driver *codec_dai_drv;
 	struct snd_soc_pcm_stream *codec_stream;
@@ -388,7 +391,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		cpu_stream = &cpu_dai_drv->capture;
 
 	/* first calculate min/max only for CODECs in the DAI link */
-	for (i = 0; i < rtd->num_codecs; i++) {
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 
 		/*
 		 * Skip CODECs which don't support the current stream type.
@@ -399,11 +402,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
 		 * bailed out on a higher level, since there would be no
 		 * CODEC to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(rtd->codec_dais[i],
+		if (!snd_soc_dai_stream_valid(codec_dai,
 					      substream->stream))
 			continue;
 
-		codec_dai_drv = rtd->codec_dais[i]->driver;
+		codec_dai_drv = codec_dai->driver;
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			codec_stream = &codec_dai_drv->playback;
 		else
@@ -482,8 +485,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	int i, ret = 0;
 
 	pinctrl_pm_select_default_state(cpu_dai->dev);
-	for (i = 0; i < rtd->num_codecs; i++)
-		pinctrl_pm_select_default_state(rtd->codec_dais[i]->dev);
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		pinctrl_pm_select_default_state(codec_dai->dev);
 
 	for_each_rtdcom(rtd, rtdcom) {
 		component = rtdcom->component;
@@ -520,8 +523,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	}
 	component = NULL;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->startup) {
 			ret = codec_dai->driver->ops->startup(substream,
 							      codec_dai);
@@ -588,10 +590,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			goto config_err;
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if (rtd->codec_dais[i]->active) {
-			ret = soc_pcm_apply_symmetry(substream,
-						     rtd->codec_dais[i]);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (codec_dai->active) {
+			ret = soc_pcm_apply_symmetry(substream, codec_dai);
 			if (ret != 0)
 				goto config_err;
 		}
@@ -620,8 +621,7 @@ machine_err:
 	i = rtd->num_codecs;
 
 codec_dai_err:
-	while (--i >= 0) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->shutdown)
 			codec_dai->driver->ops->shutdown(substream, codec_dai);
 	}
@@ -641,9 +641,9 @@ out:
 		pm_runtime_put_autosuspend(component->dev);
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if (!rtd->codec_dais[i]->active)
-			pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (!codec_dai->active)
+			pinctrl_pm_select_sleep_state(codec_dai->dev);
 	}
 	if (!cpu_dai->active)
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -701,8 +701,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (!cpu_dai->active)
 		cpu_dai->rate = 0;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (!codec_dai->active)
 			codec_dai->rate = 0;
 	}
@@ -712,8 +711,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (cpu_dai->driver->ops->shutdown)
 		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->shutdown)
 			codec_dai->driver->ops->shutdown(substream, codec_dai);
 	}
@@ -751,9 +749,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 		pm_runtime_put_autosuspend(component->dev);
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if (!rtd->codec_dais[i]->active)
-			pinctrl_pm_select_sleep_state(rtd->codec_dais[i]->dev);
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if (!codec_dai->active)
+			pinctrl_pm_select_sleep_state(codec_dai->dev);
 	}
 	if (!cpu_dai->active)
 		pinctrl_pm_select_sleep_state(cpu_dai->dev);
@@ -801,8 +799,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->prepare) {
 			ret = codec_dai->driver->ops->prepare(substream,
 							      codec_dai);
@@ -834,8 +831,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	snd_soc_dapm_stream_event(rtd, substream->stream,
 			SND_SOC_DAPM_STREAM_START);
 
-	for (i = 0; i < rtd->num_codecs; i++)
-		snd_soc_dai_digital_mute(rtd->codec_dais[i], 0,
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		snd_soc_dai_digital_mute(codec_dai, 0,
 					 substream->stream);
 	snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
 
@@ -920,6 +917,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component;
 	struct snd_soc_rtdcom_list *rtdcom;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai;
 	int i, ret = 0;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -932,8 +930,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		struct snd_pcm_hw_params codec_params;
 
 		/*
@@ -1018,8 +1015,7 @@ interface_err:
 	i = rtd->num_codecs;
 
 codec_err:
-	while (--i >= 0) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->hw_free)
 			codec_dai->driver->ops->hw_free(substream, codec_dai);
 		codec_dai->rate = 0;
@@ -1052,8 +1048,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		cpu_dai->sample_bits = 0;
 	}
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->active == 1) {
 			codec_dai->rate = 0;
 			codec_dai->channels = 0;
@@ -1062,10 +1057,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	}
 
 	/* apply codec digital mute */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		if ((playback && rtd->codec_dais[i]->playback_active == 1) ||
-		    (!playback && rtd->codec_dais[i]->capture_active == 1))
-			snd_soc_dai_digital_mute(rtd->codec_dais[i], 1,
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		if ((playback && codec_dai->playback_active == 1) ||
+		    (!playback && codec_dai->capture_active == 1))
+			snd_soc_dai_digital_mute(codec_dai, 1,
 						 substream->stream);
 	}
 
@@ -1077,8 +1072,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	soc_pcm_components_hw_free(substream, NULL);
 
 	/* now free hw params for the DAIs  */
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->hw_free)
 			codec_dai->driver->ops->hw_free(substream, codec_dai);
 	}
@@ -1099,8 +1093,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->trigger) {
 			ret = codec_dai->driver->ops->trigger(substream,
 							      cmd, codec_dai);
@@ -1144,8 +1137,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->bespoke_trigger) {
 			ret = codec_dai->driver->ops->bespoke_trigger(substream,
 								cmd, codec_dai);
@@ -1199,8 +1191,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	if (cpu_dai->driver->ops->delay)
 		delay += cpu_dai->driver->ops->delay(substream, cpu_dai);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dai(rtd, i, codec_dai) {
 		if (codec_dai->driver->ops->delay)
 			codec_delay = max(codec_delay,
 					codec_dai->driver->ops->delay(substream,
@@ -1220,7 +1211,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 
 	/* only add new dpcms */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		if (dpcm->be == be && dpcm->fe == fe)
 			return 0;
 	}
@@ -1261,7 +1252,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
 
 	be_substream = snd_soc_dpcm_get_substream(be, stream);
 
-	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+	for_each_dpcm_fe(be, stream, dpcm) {
 		if (dpcm->fe == fe)
 			continue;
 
@@ -1281,7 +1272,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm, *d;
 
-	list_for_each_entry_safe(dpcm, d, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be_safe(fe, stream, dpcm, d) {
 		dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n",
 				stream ? "capture" : "playback",
 				dpcm->be->dai_link->name);
@@ -1310,12 +1301,13 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 		struct snd_soc_dapm_widget *widget, int stream)
 {
 	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_dai *dai;
 	int i;
 
 	dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		list_for_each_entry(be, &card->rtd_list, list) {
+		for_each_card_rtds(card, be) {
 
 			if (!be->dai_link->no_pcm)
 				continue;
@@ -1327,15 +1319,14 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 			if (be->cpu_dai->playback_widget == widget)
 				return be;
 
-			for (i = 0; i < be->num_codecs; i++) {
-				struct snd_soc_dai *dai = be->codec_dais[i];
+			for_each_rtd_codec_dai(be, i, dai) {
 				if (dai->playback_widget == widget)
 					return be;
 			}
 		}
 	} else {
 
-		list_for_each_entry(be, &card->rtd_list, list) {
+		for_each_card_rtds(card, be) {
 
 			if (!be->dai_link->no_pcm)
 				continue;
@@ -1347,8 +1338,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 			if (be->cpu_dai->capture_widget == widget)
 				return be;
 
-			for (i = 0; i < be->num_codecs; i++) {
-				struct snd_soc_dai *dai = be->codec_dais[i];
+			for_each_rtd_codec_dai(be, i, dai) {
 				if (dai->capture_widget == widget)
 					return be;
 			}
@@ -1388,32 +1378,31 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
 {
 	struct snd_soc_card *card = widget->dapm->card;
 	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai *dai;
 	int i;
 
 	if (dir == SND_SOC_DAPM_DIR_OUT) {
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+		for_each_card_rtds(card, rtd) {
 			if (!rtd->dai_link->no_pcm)
 				continue;
 
 			if (rtd->cpu_dai->playback_widget == widget)
 				return true;
 
-			for (i = 0; i < rtd->num_codecs; ++i) {
-				struct snd_soc_dai *dai = rtd->codec_dais[i];
+			for_each_rtd_codec_dai(rtd, i, dai) {
 				if (dai->playback_widget == widget)
 					return true;
 			}
 		}
 	} else { /* SND_SOC_DAPM_DIR_IN */
-		list_for_each_entry(rtd, &card->rtd_list, list) {
+		for_each_card_rtds(card, rtd) {
 			if (!rtd->dai_link->no_pcm)
 				continue;
 
 			if (rtd->cpu_dai->capture_widget == widget)
 				return true;
 
-			for (i = 0; i < rtd->num_codecs; ++i) {
-				struct snd_soc_dai *dai = rtd->codec_dais[i];
+			for_each_rtd_codec_dai(rtd, i, dai) {
 				if (dai->capture_widget == widget)
 					return true;
 			}
@@ -1445,10 +1434,11 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
 	struct snd_soc_dpcm *dpcm;
 	struct snd_soc_dapm_widget_list *list = *list_;
 	struct snd_soc_dapm_widget *widget;
+	struct snd_soc_dai *dai;
 	int prune = 0;
 
 	/* Destroy any old FE <--> BE connections */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		unsigned int i;
 
 		/* is there a valid CPU DAI widget for this BE */
@@ -1459,8 +1449,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
 			continue;
 
 		/* is there a valid CODEC DAI widget for this BE */
-		for (i = 0; i < dpcm->be->num_codecs; i++) {
-			struct snd_soc_dai *dai = dpcm->be->codec_dais[i];
+		for_each_rtd_codec_dai(dpcm->be, i, dai) {
 			widget = dai_get_widget(dai, stream);
 
 			/* prune the BE if it's no longer in our active list */
@@ -1555,7 +1544,7 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+	for_each_dpcm_be(fe, stream, dpcm)
 		dpcm->be->dpcm[stream].runtime_update =
 						SND_SOC_DPCM_UPDATE_NO;
 }
@@ -1566,7 +1555,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 
 	/* disable any enabled and non active backends */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -1595,7 +1584,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
 	int err, count = 0;
 
 	/* only startup BE DAIs that are either sinks or sources to this FE DAI */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -1649,7 +1638,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
 
 unwind:
 	/* disable any enabled and non active backends */
-	list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be_rollback(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
@@ -1680,7 +1669,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 				 struct snd_soc_pcm_stream *stream)
 {
 	runtime->hw.rate_min = stream->rate_min;
-	runtime->hw.rate_max = stream->rate_max;
+	runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX);
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
 	if (runtime->hw.formats)
@@ -1695,6 +1684,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *fe = substream->private_data;
 	struct snd_soc_dpcm *dpcm;
+	struct snd_soc_dai *dai;
 	int stream = substream->stream;
 
 	if (!fe->dai_link->dpcm_merged_format)
@@ -1705,22 +1695,21 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream,
 	 * if FE want to use it (= dpcm_merged_format)
 	 */
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_soc_dai_driver *codec_dai_drv;
 		struct snd_soc_pcm_stream *codec_stream;
 		int i;
 
-		for (i = 0; i < be->num_codecs; i++) {
+		for_each_rtd_codec_dai(be, i, dai) {
 			/*
 			 * Skip CODECs which don't support the current stream
 			 * type. See soc_pcm_init_runtime_hw() for more details
 			 */
-			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
-						      stream))
+			if (!snd_soc_dai_stream_valid(dai, stream))
 				continue;
 
-			codec_dai_drv = be->codec_dais[i]->driver;
+			codec_dai_drv = dai->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
 			else
@@ -1747,7 +1736,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
 	 * if FE want to use it (= dpcm_merged_chan)
 	 */
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
 		struct snd_soc_dai_driver *codec_dai_drv;
@@ -1799,12 +1788,13 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
 	 * if FE want to use it (= dpcm_merged_chan)
 	 */
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
 		struct snd_soc_dai_driver *codec_dai_drv;
 		struct snd_soc_pcm_stream *codec_stream;
 		struct snd_soc_pcm_stream *cpu_stream;
+		struct snd_soc_dai *dai;
 		int i;
 
 		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1816,16 +1806,15 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream,
 		*rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
 		*rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
 
-		for (i = 0; i < be->num_codecs; i++) {
+		for_each_rtd_codec_dai(be, i, dai) {
 			/*
 			 * Skip CODECs which don't support the current stream
 			 * type. See soc_pcm_init_runtime_hw() for more details
 			 */
-			if (!snd_soc_dai_stream_valid(be->codec_dais[i],
-						      stream))
+			if (!snd_soc_dai_stream_valid(dai, stream))
 				continue;
 
-			codec_dai_drv = be->codec_dais[i]->driver;
+			codec_dai_drv = dai->driver;
 			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 				codec_stream = &codec_dai_drv->playback;
 			else
@@ -1902,11 +1891,12 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
 	}
 
 	/* apply symmetry for BE */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
 		struct snd_soc_pcm_runtime *rtd = be_substream->private_data;
+		struct snd_soc_dai *codec_dai;
 		int i;
 
 		if (rtd->dai_link->be_hw_params_fixup)
@@ -1923,10 +1913,10 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream,
 				return err;
 		}
 
-		for (i = 0; i < rtd->num_codecs; i++) {
-			if (rtd->codec_dais[i]->active) {
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
+			if (codec_dai->active) {
 				err = soc_pcm_apply_symmetry(fe_substream,
-							     rtd->codec_dais[i]);
+							     codec_dai);
 				if (err < 0)
 					return err;
 			}
@@ -1986,7 +1976,7 @@ int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
 	struct snd_soc_dpcm *dpcm;
 
 	/* only shutdown BEs that are either sinks or sources to this FE DAI */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2050,7 +2040,7 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 
 	/* only hw_params backends that are either sinks or sources
 	 * to this frontend DAI */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2119,7 +2109,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
 	struct snd_soc_dpcm *dpcm;
 	int ret;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2170,7 +2160,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
 
 unwind:
 	/* disable any enabled and non active backends */
-	list_for_each_entry_continue_reverse(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be_rollback(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
@@ -2250,7 +2240,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 	struct snd_soc_dpcm *dpcm;
 	int ret = 0;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2436,7 +2426,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
 	struct snd_soc_dpcm *dpcm;
 	int ret = 0;
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *be_substream =
@@ -2646,7 +2636,7 @@ close:
 	dpcm_be_dai_shutdown(fe, stream);
 disconnect:
 	/* disconnect any non started BEs */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
 				dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
@@ -2771,14 +2761,14 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 
 	mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
 	/* shutdown all old paths first */
-	list_for_each_entry(fe, &card->rtd_list, list) {
+	for_each_card_rtds(card, fe) {
 		ret = soc_dpcm_fe_runtime_update(fe, 0);
 		if (ret)
 			goto out;
 	}
 
 	/* bring new paths up */
-	list_for_each_entry(fe, &card->rtd_list, list) {
+	for_each_card_rtds(card, fe) {
 		ret = soc_dpcm_fe_runtime_update(fe, 1);
 		if (ret)
 			goto out;
@@ -2791,10 +2781,9 @@ out:
 int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 {
 	struct snd_soc_dpcm *dpcm;
-	struct list_head *clients =
-		&fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients;
+	struct snd_soc_dai *dai;
 
-	list_for_each_entry(dpcm, clients, list_be) {
+	for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
 
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		int i;
@@ -2802,8 +2791,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
 		if (be->dai_link->ignore_suspend)
 			continue;
 
-		for (i = 0; i < be->num_codecs; i++) {
-			struct snd_soc_dai *dai = be->codec_dais[i];
+		for_each_rtd_codec_dai(be, i, dai) {
 			struct snd_soc_dai_driver *drv = dai->driver;
 
 			dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
@@ -2844,7 +2832,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 	ret = dpcm_fe_dai_startup(fe_substream);
 	if (ret < 0) {
 		/* clean up all links */
-		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		for_each_dpcm_be(fe, stream, dpcm)
 			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 		dpcm_be_disconnect(fe, stream);
@@ -2867,7 +2855,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
 	ret = dpcm_fe_dai_shutdown(fe_substream);
 
 	/* mark FE's links ready to prune */
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+	for_each_dpcm_be(fe, stream, dpcm)
 		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
 	dpcm_be_disconnect(fe, stream);
@@ -3041,8 +3029,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		playback = rtd->dai_link->dpcm_playback;
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
-		for (i = 0; i < rtd->num_codecs; i++) {
-			codec_dai = rtd->codec_dais[i];
+		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (codec_dai->driver->playback.channels_min)
 				playback = 1;
 			if (codec_dai->driver->capture.channels_min)
@@ -3230,7 +3217,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 	int state;
 
-	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+	for_each_dpcm_fe(be, stream, dpcm) {
 
 		if (dpcm->fe == fe)
 			continue;
@@ -3257,7 +3244,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 	struct snd_soc_dpcm *dpcm;
 	int state;
 
-	list_for_each_entry(dpcm, &be->dpcm[stream].fe_clients, list_fe) {
+	for_each_dpcm_fe(be, stream, dpcm) {
 
 		if (dpcm->fe == fe)
 			continue;
@@ -3337,7 +3324,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
 		goto out;
 	}
 
-	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		params = &dpcm->hw_params;
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 66e77e020745..045ef136903d 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -993,7 +993,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count,
 				kfree(se);
 				continue;
 			}
-			/* fall through and create texts */
+			/* fall through */
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1310,7 +1310,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
 					ec->hdr.name);
 				goto err_se;
 			}
-			/* fall through to create texts */
+			/* fall through */
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
@@ -1565,17 +1565,6 @@ widget:
 		widget = snd_soc_dapm_new_control_unlocked(dapm, &template);
 	if (IS_ERR(widget)) {
 		ret = PTR_ERR(widget);
-		/* Do not nag about probe deferrals */
-		if (ret != -EPROBE_DEFER)
-			dev_err(tplg->dev,
-				"ASoC: failed to create widget %s controls (%d)\n",
-				w->name, ret);
-		goto hdr_err;
-	}
-	if (widget == NULL) {
-		dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n",
-			w->name);
-		ret = -ENOMEM;
 		goto hdr_err;
 	}
 
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index e0c93496c0cd..e3b9dd634c6d 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static const struct snd_pcm_ops dummy_dma_ops = {
+static const struct snd_pcm_ops snd_dummy_dma_ops = {
 	.open		= dummy_dma_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 };
 
 static const struct snd_soc_component_driver dummy_platform = {
-	.ops = &dummy_dma_ops,
+	.ops = &snd_dummy_dma_ops,
 };
 
 static const struct snd_soc_component_driver dummy_codec = {
diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c
index f22654253c43..d597eba61992 100644
--- a/sound/soc/stm/stm32_sai.c
+++ b/sound/soc/stm/stm32_sai.c
@@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client,
 
 	if (!pdev) {
 		dev_err(&sai_client->pdev->dev,
-			"Device not found for node %s\n", np_provider->name);
+			"Device not found for node %pOFn\n", np_provider);
 		return -ENODEV;
 	}
 
diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h
index f25422174909..08de899c766b 100644
--- a/sound/soc/stm/stm32_sai.h
+++ b/sound/soc/stm/stm32_sai.h
@@ -91,6 +91,9 @@
 #define SAI_XCR1_OSR_SHIFT	26
 #define SAI_XCR1_OSR		BIT(SAI_XCR1_OSR_SHIFT)
 
+#define SAI_XCR1_MCKEN_SHIFT	27
+#define SAI_XCR1_MCKEN		BIT(SAI_XCR1_MCKEN_SHIFT)
+
 /******************* Bit definition for SAI_XCR2 register *******************/
 #define SAI_XCR2_FTH_SHIFT	0
 #define SAI_XCR2_FTH_MASK	GENMASK(2, SAI_XCR2_FTH_SHIFT)
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 06fba9650ac4..31d22abd3204 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -17,6 +17,7 @@
  */
 
 #include <linux/clk.h>
+#include <linux/clk-provider.h>
 #include <linux/kernel.h>
 #include <linux/module.h>
 #include <linux/of_irq.h>
@@ -68,6 +69,8 @@
 #define SAI_IEC60958_BLOCK_FRAMES	192
 #define SAI_IEC60958_STATUS_BYTES	24
 
+#define SAI_MCLK_NAME_LEN		32
+
 /**
  * struct stm32_sai_sub_data - private data of SAI sub block (block A or B)
  * @pdev: device data pointer
@@ -80,6 +83,7 @@
  * @pdata: SAI block parent data pointer
  * @np_sync_provider: synchronization provider node
  * @sai_ck: kernel clock feeding the SAI clock generator
+ * @sai_mclk: master clock from SAI mclk provider
  * @phys_addr: SAI registers physical base address
  * @mclk_rate: SAI block master clock frequency (Hz). set at init
  * @id: SAI sub block id corresponding to sub-block A or B
@@ -110,6 +114,7 @@ struct stm32_sai_sub_data {
 	struct stm32_sai_data *pdata;
 	struct device_node *np_sync_provider;
 	struct clk *sai_ck;
+	struct clk *sai_mclk;
 	dma_addr_t phys_addr;
 	unsigned int mclk_rate;
 	unsigned int id;
@@ -251,6 +256,177 @@ static const struct snd_kcontrol_new iec958_ctls = {
 	.put = snd_pcm_iec958_put,
 };
 
+struct stm32_sai_mclk_data {
+	struct clk_hw hw;
+	unsigned long freq;
+	struct stm32_sai_sub_data *sai_data;
+};
+
+#define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw)
+#define STM32_SAI_MAX_CLKS 1
+
+static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai,
+				 unsigned long input_rate,
+				 unsigned long output_rate)
+{
+	int version = sai->pdata->conf->version;
+	int div;
+
+	div = DIV_ROUND_CLOSEST(input_rate, output_rate);
+	if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+		dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+		return -EINVAL;
+	}
+	dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div);
+
+	if (input_rate % div)
+		dev_dbg(&sai->pdev->dev,
+			"Rate not accurate. requested (%ld), actual (%ld)\n",
+			output_rate, input_rate / div);
+
+	return div;
+}
+
+static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai,
+				 unsigned int div)
+{
+	int version = sai->pdata->conf->version;
+	int ret, cr1, mask;
+
+	if (div > SAI_XCR1_MCKDIV_MAX(version)) {
+		dev_err(&sai->pdev->dev, "Divider %d out of range\n", div);
+		return -EINVAL;
+	}
+
+	mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
+	cr1 = SAI_XCR1_MCKDIV_SET(div);
+	ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
+	if (ret < 0)
+		dev_err(&sai->pdev->dev, "Failed to update CR1 register\n");
+
+	return ret;
+}
+
+static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate,
+				      unsigned long *prate)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+	int div;
+
+	div = stm32_sai_get_clk_div(sai, *prate, rate);
+	if (div < 0)
+		return div;
+
+	mclk->freq = *prate / div;
+
+	return mclk->freq;
+}
+
+static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw,
+						unsigned long parent_rate)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+
+	return mclk->freq;
+}
+
+static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate,
+				   unsigned long parent_rate)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+	unsigned int div;
+	int ret;
+
+	div = stm32_sai_get_clk_div(sai, parent_rate, rate);
+	if (div < 0)
+		return div;
+
+	ret = stm32_sai_set_clk_div(sai, div);
+	if (ret)
+		return ret;
+
+	mclk->freq = rate;
+
+	return 0;
+}
+
+static int stm32_sai_mclk_enable(struct clk_hw *hw)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+	dev_dbg(&sai->pdev->dev, "Enable master clock\n");
+
+	return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
+				  SAI_XCR1_MCKEN, SAI_XCR1_MCKEN);
+}
+
+static void stm32_sai_mclk_disable(struct clk_hw *hw)
+{
+	struct stm32_sai_mclk_data *mclk = to_mclk_data(hw);
+	struct stm32_sai_sub_data *sai = mclk->sai_data;
+
+	dev_dbg(&sai->pdev->dev, "Disable master clock\n");
+
+	regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0);
+}
+
+static const struct clk_ops mclk_ops = {
+	.enable = stm32_sai_mclk_enable,
+	.disable = stm32_sai_mclk_disable,
+	.recalc_rate = stm32_sai_mclk_recalc_rate,
+	.round_rate = stm32_sai_mclk_round_rate,
+	.set_rate = stm32_sai_mclk_set_rate,
+};
+
+static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
+{
+	struct clk_hw *hw;
+	struct stm32_sai_mclk_data *mclk;
+	struct device *dev = &sai->pdev->dev;
+	const char *pname = __clk_get_name(sai->sai_ck);
+	char *mclk_name, *p, *s = (char *)pname;
+	int ret, i = 0;
+
+	mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+	if (!mclk)
+		return -ENOMEM;
+
+	mclk_name = devm_kcalloc(dev, sizeof(char),
+				 SAI_MCLK_NAME_LEN, GFP_KERNEL);
+	if (!mclk_name)
+		return -ENOMEM;
+
+	/*
+	 * Forge mclk clock name from parent clock name and suffix.
+	 * String after "_" char is stripped in parent name.
+	 */
+	p = mclk_name;
+	while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) {
+		*p++ = *s++;
+		i++;
+	}
+	STM_SAI_IS_SUB_A(sai) ?
+		strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6);
+
+	mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0);
+	mclk->sai_data = sai;
+	hw = &mclk->hw;
+
+	dev_dbg(dev, "Register master clock %s\n", mclk_name);
+	ret = devm_clk_hw_register(&sai->pdev->dev, hw);
+	if (ret) {
+		dev_err(dev, "mclk register returned %d\n", ret);
+		return ret;
+	}
+	sai->sai_mclk = hw->clk;
+
+	/* register mclk provider */
+	return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw);
+}
+
 static irqreturn_t stm32_sai_isr(int irq, void *devid)
 {
 	struct stm32_sai_sub_data *sai = (struct stm32_sai_sub_data *)devid;
@@ -312,15 +488,25 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai,
 	struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	int ret;
 
-	if ((dir == SND_SOC_CLOCK_OUT) && sai->master) {
+	if (dir == SND_SOC_CLOCK_OUT) {
 		ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
 					 SAI_XCR1_NODIV,
 					 (unsigned int)~SAI_XCR1_NODIV);
 		if (ret < 0)
 			return ret;
 
-		sai->mclk_rate = freq;
 		dev_dbg(cpu_dai->dev, "SAI MCLK frequency is %uHz\n", freq);
+		sai->mclk_rate = freq;
+
+		if (sai->sai_mclk) {
+			ret = clk_set_rate_exclusive(sai->sai_mclk,
+						     sai->mclk_rate);
+			if (ret) {
+				dev_err(cpu_dai->dev,
+					"Could not set mclk rate\n");
+				return ret;
+			}
+		}
 	}
 
 	return 0;
@@ -715,15 +901,9 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 {
 	struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	int cr1, mask, div = 0;
-	int sai_clk_rate, mclk_ratio, den, ret;
-	int version = sai->pdata->conf->version;
+	int sai_clk_rate, mclk_ratio, den;
 	unsigned int rate = params_rate(params);
 
-	if (!sai->mclk_rate) {
-		dev_err(cpu_dai->dev, "Mclk rate is null\n");
-		return -EINVAL;
-	}
-
 	if (!(rate % 11025))
 		clk_set_parent(sai->sai_ck, sai->pdata->clk_x11k);
 	else
@@ -731,14 +911,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 	sai_clk_rate = clk_get_rate(sai->sai_ck);
 
 	if (STM_SAI_IS_F4(sai->pdata)) {
-		/*
-		 * mclk_rate = 256 * fs
-		 * MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
-		 * MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+		/* mclk on (NODIV=0)
+		 *   mclk_rate = 256 * fs
+		 *   MCKDIV = 0 if sai_ck < 3/2 * mclk_rate
+		 *   MCKDIV = sai_ck / (2 * mclk_rate) otherwise
+		 * mclk off (NODIV=1)
+		 *   MCKDIV ignored. sck = sai_ck
 		 */
-		if (2 * sai_clk_rate >= 3 * sai->mclk_rate)
-			div = DIV_ROUND_CLOSEST(sai_clk_rate,
-						2 * sai->mclk_rate);
+		if (!sai->mclk_rate)
+			return 0;
+
+		if (2 * sai_clk_rate >= 3 * sai->mclk_rate) {
+			div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+						    2 * sai->mclk_rate);
+			if (div < 0)
+				return div;
+		}
 	} else {
 		/*
 		 * TDM mode :
@@ -750,8 +938,10 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 		 * Note: NOMCK/NODIV correspond to same bit.
 		 */
 		if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
-			div = DIV_ROUND_CLOSEST(sai_clk_rate,
-						(params_rate(params) * 128));
+			div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+						    rate * 128);
+			if (div < 0)
+				return div;
 		} else {
 			if (sai->mclk_rate) {
 				mclk_ratio = sai->mclk_rate / rate;
@@ -764,31 +954,22 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai,
 						mclk_ratio);
 					return -EINVAL;
 				}
-				div = DIV_ROUND_CLOSEST(sai_clk_rate,
-							sai->mclk_rate);
+				div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+							    sai->mclk_rate);
+				if (div < 0)
+					return div;
 			} else {
 				/* mclk-fs not set, master clock not active */
 				den = sai->fs_length * params_rate(params);
-				div = DIV_ROUND_CLOSEST(sai_clk_rate, den);
+				div = stm32_sai_get_clk_div(sai, sai_clk_rate,
+							    den);
+				if (div < 0)
+					return div;
 			}
 		}
 	}
 
-	if (div > SAI_XCR1_MCKDIV_MAX(version)) {
-		dev_err(cpu_dai->dev, "Divider %d out of range\n", div);
-		return -EINVAL;
-	}
-	dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div);
-
-	mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
-	cr1 = SAI_XCR1_MCKDIV_SET(div);
-	ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
-	if (ret < 0) {
-		dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
-		return ret;
-	}
-
-	return 0;
+	return stm32_sai_set_clk_div(sai, div);
 }
 
 static int stm32_sai_hw_params(struct snd_pcm_substream *substream,
@@ -881,6 +1062,9 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream,
 			   SAI_XCR1_NODIV);
 
 	clk_disable_unprepare(sai->sai_ck);
+
+	clk_rate_exclusive_put(sai->sai_mclk);
+
 	sai->substream = NULL;
 }
 
@@ -903,6 +1087,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
 	int cr1 = 0, cr1_mask;
 
+	sai->cpu_dai = cpu_dai;
+
 	sai->dma_params.addr = (dma_addr_t)(sai->phys_addr + STM_SAI_DR_REGX);
 	/*
 	 * DMA supports 4, 8 or 16 burst sizes. Burst size 4 is the best choice,
@@ -1124,16 +1310,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
 	sai->sync = SAI_SYNC_NONE;
 	if (args.np) {
 		if (args.np == np) {
-			dev_err(&pdev->dev, "%s sync own reference\n",
-				np->name);
+			dev_err(&pdev->dev, "%pOFn sync own reference\n", np);
 			of_node_put(args.np);
 			return -EINVAL;
 		}
 
 		sai->np_sync_provider  = of_get_parent(args.np);
 		if (!sai->np_sync_provider) {
-			dev_err(&pdev->dev, "%s parent node not found\n",
-				np->name);
+			dev_err(&pdev->dev, "%pOFn parent node not found\n",
+				np);
 			of_node_put(args.np);
 			return -ENODEV;
 		}
@@ -1182,6 +1367,23 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev,
 		return PTR_ERR(sai->sai_ck);
 	}
 
+	if (STM_SAI_IS_F4(sai->pdata))
+		return 0;
+
+	/* Register mclk provider if requested */
+	if (of_find_property(np, "#clock-cells", NULL)) {
+		ret = stm32_sai_add_mclk_provider(sai);
+		if (ret < 0)
+			return ret;
+	} else {
+		sai->sai_mclk = devm_clk_get(&pdev->dev, "MCLK");
+		if (IS_ERR(sai->sai_mclk)) {
+			if (PTR_ERR(sai->sai_mclk) != -ENOENT)
+				return PTR_ERR(sai->sai_mclk);
+			sai->sai_mclk = NULL;
+		}
+	}
+
 	return 0;
 }
 
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
index 22408bc2d6ec..66aad0d3f9c7 100644
--- a/sound/soc/sunxi/Kconfig
+++ b/sound/soc/sunxi/Kconfig
@@ -12,7 +12,7 @@ config SND_SUN4I_CODEC
 config SND_SUN8I_CODEC
 	tristate "Allwinner SUN8I audio codec"
 	depends on OF
-	depends on MACH_SUN8I || COMPILE_TEST
+	depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
 	select REGMAP_MMIO
 	help
 	  This option enables the digital part of the internal audio codec for
@@ -23,11 +23,19 @@ config SND_SUN8I_CODEC
 config SND_SUN8I_CODEC_ANALOG
 	tristate "Allwinner sun8i Codec Analog Controls Support"
 	depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST
-	select REGMAP
+	select SND_SUN8I_ADDA_PR_REGMAP
 	help
 	  Say Y or M if you want to add support for the analog controls for
 	  the codec embedded in newer Allwinner SoCs.
 
+config SND_SUN50I_CODEC_ANALOG
+	tristate "Allwinner sun50i Codec Analog Controls Support"
+	depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
+	select SND_SUNXI_ADDA_PR_REGMAP
+	help
+	  Say Y or M if you want to add support for the analog controls for
+	  the codec embedded in Allwinner A64 SoC.
+
 config SND_SUN4I_I2S
 	tristate "Allwinner A10 I2S Support"
 	select SND_SOC_GENERIC_DMAENGINE_PCM
@@ -45,4 +53,9 @@ config SND_SUN4I_SPDIF
 	help
 	  Say Y or M to add support for the S/PDIF audio block in the Allwinner
 	  A10 and affiliated SoCs.
+
+config SND_SUN8I_ADDA_PR_REGMAP
+	tristate
+	select REGMAP
+
 endmenu
diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile
index 4a9ef67386ca..a86be340a076 100644
--- a/sound/soc/sunxi/Makefile
+++ b/sound/soc/sunxi/Makefile
@@ -3,4 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
 obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o
 obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o
 obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o
+obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o
 obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o
+obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index a4aa931ebfae..d5ec1a20499d 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-static int sun4i_i2s_startup(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-	/* Enable the whole hardware block */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
-
-	/* Enable the first output line */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_SDO_EN_MASK,
-			   SUN4I_I2S_CTRL_SDO_EN(0));
-
-
-	return clk_prepare_enable(i2s->mod_clk);
-}
-
-static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
-{
-	struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-
-	clk_disable_unprepare(i2s->mod_clk);
-
-	/* Disable our output lines */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
-
-	/* Disable the whole hardware block */
-	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
-			   SUN4I_I2S_CTRL_GL_EN, 0);
-}
-
 static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 				unsigned int freq, int dir)
 {
@@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
 	.hw_params	= sun4i_i2s_hw_params,
 	.set_fmt	= sun4i_i2s_set_fmt,
 	.set_sysclk	= sun4i_i2s_set_sysclk,
-	.shutdown	= sun4i_i2s_shutdown,
-	.startup	= sun4i_i2s_startup,
 	.trigger	= sun4i_i2s_trigger,
 };
 
@@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev)
 		goto err_disable_clk;
 	}
 
+	/* Enable the whole hardware block */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN);
+
+	/* Enable the first output line */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_SDO_EN_MASK,
+			   SUN4I_I2S_CTRL_SDO_EN(0));
+
+	ret = clk_prepare_enable(i2s->mod_clk);
+	if (ret) {
+		dev_err(dev, "Failed to enable module clock\n");
+		goto err_disable_clk;
+	}
+
 	return 0;
 
 err_disable_clk:
@@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev)
 {
 	struct sun4i_i2s *i2s = dev_get_drvdata(dev);
 
+	clk_disable_unprepare(i2s->mod_clk);
+
+	/* Disable our output lines */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_SDO_EN_MASK, 0);
+
+	/* Disable the whole hardware block */
+	regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
+			   SUN4I_I2S_CTRL_GL_EN, 0);
+
 	regcache_cache_only(i2s->regmap, true);
 
 	clk_disable_unprepare(i2s->bus_clk);
@@ -961,6 +950,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = {
 	.field_rxchansel	= REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2),
 };
 
+static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
+	.has_reset		= true,
+	.reg_offset_txdata	= SUN8I_I2S_FIFO_TX_REG,
+	.sun4i_i2s_regmap	= &sun4i_i2s_regmap_config,
+	.has_slave_select_bit	= true,
+	.field_clkdiv_mclk_en	= REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7),
+	.field_fmt_wss		= REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3),
+	.field_fmt_sr		= REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5),
+	.field_fmt_bclk		= REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6),
+	.field_fmt_lrclk	= REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7),
+	.field_fmt_mode		= REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1),
+	.field_txchanmap	= REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31),
+	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
+	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
+	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
+};
+
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
 					struct sun4i_i2s *i2s)
 {
@@ -1169,6 +1175,10 @@ static const struct of_device_id sun4i_i2s_match[] = {
 		.compatible = "allwinner,sun8i-h3-i2s",
 		.data = &sun8i_h3_i2s_quirks,
 	},
+	{
+		.compatible = "allwinner,sun50i-a64-codec-i2s",
+		.data = &sun50i_a64_codec_i2s_quirks,
+	},
 	{}
 };
 MODULE_DEVICE_TABLE(of, sun4i_i2s_match);
diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c
new file mode 100644
index 000000000000..8f5f999df631
--- /dev/null
+++ b/sound/soc/sunxi/sun50i-codec-analog.c
@@ -0,0 +1,444 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver supports the analog controls for the internal codec
+ * found in Allwinner's A64 SoC.
+ *
+ * Copyright (C) 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2017 Marcus Cooper <codekipper@gmail.com>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on sun8i-codec-analog.c
+ *
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Codec analog control register offsets and bit fields */
+#define SUN50I_ADDA_HP_CTRL		0x00
+#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE		7
+#define SUN50I_ADDA_HP_CTRL_HPPA_EN		6
+#define SUN50I_ADDA_HP_CTRL_HPVOL		0
+
+#define SUN50I_ADDA_OL_MIX_CTRL		0x01
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC1		6
+#define SUN50I_ADDA_OL_MIX_CTRL_MIC2		5
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONE		4
+#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN		3
+#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL		2
+#define SUN50I_ADDA_OL_MIX_CTRL_DACL		1
+#define SUN50I_ADDA_OL_MIX_CTRL_DACR		0
+
+#define SUN50I_ADDA_OR_MIX_CTRL		0x02
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC1		6
+#define SUN50I_ADDA_OR_MIX_CTRL_MIC2		5
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONE		4
+#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP		3
+#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR		2
+#define SUN50I_ADDA_OR_MIX_CTRL_DACR		1
+#define SUN50I_ADDA_OR_MIX_CTRL_DACL		0
+
+#define SUN50I_ADDA_LINEOUT_CTRL0	0x05
+#define SUN50I_ADDA_LINEOUT_CTRL0_LEN		7
+#define SUN50I_ADDA_LINEOUT_CTRL0_REN		6
+#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL	5
+#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL	4
+
+#define SUN50I_ADDA_LINEOUT_CTRL1	0x06
+#define SUN50I_ADDA_LINEOUT_CTRL1_VOL		0
+
+#define SUN50I_ADDA_MIC1_CTRL		0x07
+#define SUN50I_ADDA_MIC1_CTRL_MIC1G		4
+#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN		3
+#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST		0
+
+#define SUN50I_ADDA_MIC2_CTRL		0x08
+#define SUN50I_ADDA_MIC2_CTRL_MIC2G		4
+#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN		3
+#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST		0
+
+#define SUN50I_ADDA_LINEIN_CTRL		0x09
+#define SUN50I_ADDA_LINEIN_CTRL_LINEING		0
+
+#define SUN50I_ADDA_MIX_DAC_CTRL	0x0a
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN	7
+#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN	6
+#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN		5
+#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN		4
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE	3
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE	2
+#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS		1
+#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS		0
+
+#define SUN50I_ADDA_L_ADCMIX_SRC	0x0b
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1		6
+#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2		5
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE		4
+#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN		3
+#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL	2
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL		1
+#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR		0
+
+#define SUN50I_ADDA_R_ADCMIX_SRC	0x0c
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1		6
+#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2		5
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE		4
+#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP		3
+#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR	2
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR		1
+#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL		0
+
+#define SUN50I_ADDA_ADC_CTRL		0x0d
+#define SUN50I_ADDA_ADC_CTRL_ADCREN		7
+#define SUN50I_ADDA_ADC_CTRL_ADCLEN		6
+#define SUN50I_ADDA_ADC_CTRL_ADCG		0
+
+#define SUN50I_ADDA_HS_MBIAS_CTRL	0x0e
+#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN	7
+
+#define SUN50I_ADDA_JACK_MIC_CTRL	0x1d
+#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN	5
+
+/* mixer controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = {
+	SOC_DAPM_DOUBLE_R("DAC Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0),
+	SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0),
+	SOC_DAPM_DOUBLE_R("Line In Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic1 Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic2 Playback Switch",
+			  SUN50I_ADDA_OL_MIX_CTRL,
+			  SUN50I_ADDA_OR_MIX_CTRL,
+			  SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0),
+};
+
+/* ADC mixer controls */
+static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = {
+	SOC_DAPM_DOUBLE_R("Mixer Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0),
+	SOC_DAPM_DOUBLE_R("Line In Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic1 Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0),
+	SOC_DAPM_DOUBLE_R("Mic2 Capture Switch",
+			  SUN50I_ADDA_L_ADCMIX_SRC,
+			  SUN50I_ADDA_R_ADCMIX_SRC,
+			  SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale,
+				  -450, 150, 0);
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale,
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
+);
+
+static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale,
+	0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+	2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0),
+);
+
+
+/* volume / mute controls */
+static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = {
+	SOC_SINGLE_TLV("Headphone Playback Volume",
+		       SUN50I_ADDA_HP_CTRL,
+		       SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0,
+		       sun50i_codec_hp_vol_scale),
+
+	SOC_DOUBLE("Headphone Playback Switch",
+		   SUN50I_ADDA_MIX_DAC_CTRL,
+		   SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE,
+		   SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0),
+
+	/* Mixer pre-gain */
+	SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL,
+		       SUN50I_ADDA_MIC1_CTRL_MIC1G,
+		       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+	/* Microphone Amp boost gain */
+	SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL,
+		       SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0,
+		       sun50i_codec_mic_gain_scale),
+
+	/* Mixer pre-gain */
+	SOC_SINGLE_TLV("Mic2 Playback Volume",
+		       SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G,
+		       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+	/* Microphone Amp boost gain */
+	SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL,
+		       SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0,
+		       sun50i_codec_mic_gain_scale),
+
+	/* ADC */
+	SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL,
+		       SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0,
+		       sun50i_codec_out_mixer_pregain_scale),
+
+	/* Mixer pre-gain */
+	SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL,
+		       SUN50I_ADDA_LINEIN_CTRL_LINEING,
+		       0x7, 0, sun50i_codec_out_mixer_pregain_scale),
+
+	SOC_SINGLE_TLV("Line Out Playback Volume",
+		       SUN50I_ADDA_LINEOUT_CTRL1,
+		       SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0,
+		       sun50i_codec_lineout_vol_scale),
+
+	SOC_DOUBLE("Line Out Playback Switch",
+		   SUN50I_ADDA_LINEOUT_CTRL0,
+		   SUN50I_ADDA_LINEOUT_CTRL0_LEN,
+		   SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0),
+
+};
+
+static const char * const sun50i_codec_hp_src_enum_text[] = {
+	"DAC", "Mixer",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum,
+			    SUN50I_ADDA_MIX_DAC_CTRL,
+			    SUN50I_ADDA_MIX_DAC_CTRL_LHPIS,
+			    SUN50I_ADDA_MIX_DAC_CTRL_RHPIS,
+			    sun50i_codec_hp_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_hp_src[] = {
+	SOC_DAPM_ENUM("Headphone Source Playback Route",
+		      sun50i_codec_hp_src_enum),
+};
+
+static const char * const sun50i_codec_lineout_src_enum_text[] = {
+	"Stereo", "Mono Differential",
+};
+
+static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum,
+			    SUN50I_ADDA_LINEOUT_CTRL0,
+			    SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL,
+			    SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL,
+			    sun50i_codec_lineout_src_enum_text);
+
+static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = {
+	SOC_DAPM_ENUM("Line Out Source Playback Route",
+		      sun50i_codec_lineout_src_enum),
+};
+
+static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = {
+	/* DAC */
+	SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+			 SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0),
+	SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL,
+			 SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0),
+	/* ADC */
+	SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+			 SUN50I_ADDA_ADC_CTRL_ADCLEN, 0),
+	SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL,
+			 SUN50I_ADDA_ADC_CTRL_ADCREN, 0),
+	/*
+	 * Due to this component and the codec belonging to separate DAPM
+	 * contexts, we need to manually link the above widgets to their
+	 * stream widgets at the card level.
+	 */
+
+	SND_SOC_DAPM_MUX("Headphone Source Playback Route",
+			 SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src),
+	SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL,
+			     SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("HP"),
+
+	SND_SOC_DAPM_MUX("Line Out Source Playback Route",
+			 SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src),
+	SND_SOC_DAPM_OUTPUT("LINEOUT"),
+
+	/* Microphone inputs */
+	SND_SOC_DAPM_INPUT("MIC1"),
+
+	/* Microphone Bias */
+	SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL,
+			    SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN,
+			    0, NULL, 0),
+
+	/* Mic input path */
+	SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL,
+			 SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0),
+
+	/* Microphone input */
+	SND_SOC_DAPM_INPUT("MIC2"),
+
+	/* Microphone Bias */
+	SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL,
+			    SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN,
+			    0, NULL, 0),
+
+	/* Mic input path */
+	SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL,
+			 SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0),
+
+	/* Line input */
+	SND_SOC_DAPM_INPUT("LINEIN"),
+
+	/* Mixers */
+	SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+			   SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0,
+			   sun50i_a64_codec_mixer_controls,
+			   ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL,
+			   SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0,
+			   sun50i_a64_codec_mixer_controls,
+			   ARRAY_SIZE(sun50i_a64_codec_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+			   SUN50I_ADDA_ADC_CTRL_ADCLEN, 0,
+			   sun50i_codec_adc_mixer_controls,
+			   ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL,
+			   SUN50I_ADDA_ADC_CTRL_ADCREN, 0,
+			   sun50i_codec_adc_mixer_controls,
+			   ARRAY_SIZE(sun50i_codec_adc_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = {
+	/* Left Mixer Routes */
+	{ "Left Mixer", "DAC Playback Switch", "Left DAC" },
+	{ "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" },
+	{ "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+	/* Right Mixer Routes */
+	{ "Right Mixer", "DAC Playback Switch", "Right DAC" },
+	{ "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
+
+	/* Left ADC Mixer Routes */
+	{ "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" },
+	{ "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" },
+	{ "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+	/* Right ADC Mixer Routes */
+	{ "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" },
+	{ "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" },
+	{ "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
+
+	/* ADC Routes */
+	{ "Left ADC", NULL, "Left ADC Mixer" },
+	{ "Right ADC", NULL, "Right ADC Mixer" },
+
+	/* Headphone Routes */
+	{ "Headphone Source Playback Route", "DAC", "Left DAC" },
+	{ "Headphone Source Playback Route", "DAC", "Right DAC" },
+	{ "Headphone Source Playback Route", "Mixer", "Left Mixer" },
+	{ "Headphone Source Playback Route", "Mixer", "Right Mixer" },
+	{ "Headphone Amp", NULL, "Headphone Source Playback Route" },
+	{ "HP", NULL, "Headphone Amp" },
+
+	/* Microphone Routes */
+	{ "Mic1 Amplifier", NULL, "MIC1"},
+
+	/* Microphone Routes */
+	{ "Mic2 Amplifier", NULL, "MIC2"},
+	{ "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+	{ "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
+	{ "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+	{ "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
+
+	/* Line-in Routes */
+	{ "Left Mixer", "Line In Playback Switch", "LINEIN" },
+	{ "Right Mixer", "Line In Playback Switch", "LINEIN" },
+	{ "Left ADC Mixer", "Line In Capture Switch", "LINEIN" },
+	{ "Right ADC Mixer", "Line In Capture Switch", "LINEIN" },
+
+	/* Line-out Routes */
+	{ "Line Out Source Playback Route", "Stereo", "Left Mixer" },
+	{ "Line Out Source Playback Route", "Stereo", "Right Mixer" },
+	{ "Line Out Source Playback Route", "Mono Differential", "Left Mixer" },
+	{ "Line Out Source Playback Route", "Mono Differential",
+		"Right Mixer" },
+	{ "LINEOUT", NULL, "Line Out Source Playback Route" },
+};
+
+static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = {
+	.controls		= sun50i_a64_codec_controls,
+	.num_controls		= ARRAY_SIZE(sun50i_a64_codec_controls),
+	.dapm_widgets		= sun50i_a64_codec_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(sun50i_a64_codec_widgets),
+	.dapm_routes		= sun50i_a64_codec_routes,
+	.num_dapm_routes	= ARRAY_SIZE(sun50i_a64_codec_routes),
+};
+
+static const struct of_device_id sun50i_codec_analog_of_match[] = {
+	{
+		.compatible = "allwinner,sun50i-a64-codec-analog",
+	},
+	{}
+};
+MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match);
+
+static int sun50i_codec_analog_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct regmap *regmap;
+	void __iomem *base;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(base)) {
+		dev_err(&pdev->dev, "Failed to map the registers\n");
+		return PTR_ERR(base);
+	}
+
+	regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
+	if (IS_ERR(regmap)) {
+		dev_err(&pdev->dev, "Failed to create regmap\n");
+		return PTR_ERR(regmap);
+	}
+
+	return devm_snd_soc_register_component(&pdev->dev,
+					       &sun50i_codec_analog_cmpnt_drv,
+					       NULL, 0);
+}
+
+static struct platform_driver sun50i_codec_analog_driver = {
+	.driver = {
+		.name = "sun50i-codec-analog",
+		.of_match_table = sun50i_codec_analog_of_match,
+	},
+	.probe = sun50i_codec_analog_probe,
+};
+module_platform_driver(sun50i_codec_analog_driver);
+
+MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sun50i-codec-analog");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
new file mode 100644
index 000000000000..e68ce9d2884d
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c
@@ -0,0 +1,102 @@
+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * This driver provides regmap to access to analog part of audio codec
+ * found on Allwinner A23, A31s, A33, H3 and A64 Socs
+ *
+ * Copyright 2016 Chen-Yu Tsai <wens@csie.org>
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+
+#include "sun8i-adda-pr-regmap.h"
+
+/* Analog control register access bits */
+#define ADDA_PR			0x0		/* PRCM base + 0x1c0 */
+#define ADDA_PR_RESET			BIT(28)
+#define ADDA_PR_WRITE			BIT(24)
+#define ADDA_PR_ADDR_SHIFT		16
+#define ADDA_PR_ADDR_MASK		GENMASK(4, 0)
+#define ADDA_PR_DATA_IN_SHIFT		8
+#define ADDA_PR_DATA_IN_MASK		GENMASK(7, 0)
+#define ADDA_PR_DATA_OUT_SHIFT		0
+#define ADDA_PR_DATA_OUT_MASK		GENMASK(7, 0)
+
+/* regmap access bits */
+static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
+{
+	void __iomem *base = (void __iomem *)context;
+	u32 tmp;
+
+	/* De-assert reset */
+	writel(readl(base) | ADDA_PR_RESET, base);
+
+	/* Clear write bit */
+	writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+	/* Set register address */
+	tmp = readl(base);
+	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+	writel(tmp, base);
+
+	/* Read back value */
+	*val = readl(base) & ADDA_PR_DATA_OUT_MASK;
+
+	return 0;
+}
+
+static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
+{
+	void __iomem *base = (void __iomem *)context;
+	u32 tmp;
+
+	/* De-assert reset */
+	writel(readl(base) | ADDA_PR_RESET, base);
+
+	/* Set register address */
+	tmp = readl(base);
+	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
+	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
+	writel(tmp, base);
+
+	/* Set data to write */
+	tmp = readl(base);
+	tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
+	tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
+	writel(tmp, base);
+
+	/* Set write bit to signal a write */
+	writel(readl(base) | ADDA_PR_WRITE, base);
+
+	/* Clear write bit */
+	writel(readl(base) & ~ADDA_PR_WRITE, base);
+
+	return 0;
+}
+
+static const struct regmap_config adda_pr_regmap_cfg = {
+	.name		= "adda-pr",
+	.reg_bits	= 5,
+	.reg_stride	= 1,
+	.val_bits	= 8,
+	.reg_read	= adda_reg_read,
+	.reg_write	= adda_reg_write,
+	.fast_io	= true,
+	.max_register	= 31,
+};
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+					 void __iomem *base)
+{
+	return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg);
+}
+EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init);
+
+MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver");
+MODULE_AUTHOR("Vasily Khoruzhick <anarsoul@gmail.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:sunxi-adda-pr");
diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
new file mode 100644
index 000000000000..a5ae95dfebc1
--- /dev/null
+++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ * Copyright (C) 2018 Vasily Khoruzhick <anarsoul@gmail.com>
+ */
+
+struct regmap *sun8i_adda_pr_regmap_init(struct device *dev,
+					 void __iomem *base);
diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c
index 485e79f292c4..916a46bbc1c8 100644
--- a/sound/soc/sunxi/sun8i-codec-analog.c
+++ b/sound/soc/sunxi/sun8i-codec-analog.c
@@ -27,6 +27,8 @@
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
+#include "sun8i-adda-pr-regmap.h"
+
 /* Codec analog control register offsets and bit fields */
 #define SUN8I_ADDA_HP_VOLC		0x00
 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE		7
@@ -120,81 +122,6 @@
 #define SUN8I_ADDA_ADC_AP_EN_ADCLEN		6
 #define SUN8I_ADDA_ADC_AP_EN_ADCG		0
 
-/* Analog control register access bits */
-#define ADDA_PR			0x0		/* PRCM base + 0x1c0 */
-#define ADDA_PR_RESET			BIT(28)
-#define ADDA_PR_WRITE			BIT(24)
-#define ADDA_PR_ADDR_SHIFT		16
-#define ADDA_PR_ADDR_MASK		GENMASK(4, 0)
-#define ADDA_PR_DATA_IN_SHIFT		8
-#define ADDA_PR_DATA_IN_MASK		GENMASK(7, 0)
-#define ADDA_PR_DATA_OUT_SHIFT		0
-#define ADDA_PR_DATA_OUT_MASK		GENMASK(7, 0)
-
-/* regmap access bits */
-static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
-{
-	void __iomem *base = (void __iomem *)context;
-	u32 tmp;
-
-	/* De-assert reset */
-	writel(readl(base) | ADDA_PR_RESET, base);
-
-	/* Clear write bit */
-	writel(readl(base) & ~ADDA_PR_WRITE, base);
-
-	/* Set register address */
-	tmp = readl(base);
-	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
-	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
-	writel(tmp, base);
-
-	/* Read back value */
-	*val = readl(base) & ADDA_PR_DATA_OUT_MASK;
-
-	return 0;
-}
-
-static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
-{
-	void __iomem *base = (void __iomem *)context;
-	u32 tmp;
-
-	/* De-assert reset */
-	writel(readl(base) | ADDA_PR_RESET, base);
-
-	/* Set register address */
-	tmp = readl(base);
-	tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
-	tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
-	writel(tmp, base);
-
-	/* Set data to write */
-	tmp = readl(base);
-	tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
-	tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
-	writel(tmp, base);
-
-	/* Set write bit to signal a write */
-	writel(readl(base) | ADDA_PR_WRITE, base);
-
-	/* Clear write bit */
-	writel(readl(base) & ~ADDA_PR_WRITE, base);
-
-	return 0;
-}
-
-static const struct regmap_config adda_pr_regmap_cfg = {
-	.name		= "adda-pr",
-	.reg_bits	= 5,
-	.reg_stride	= 1,
-	.val_bits	= 8,
-	.reg_read	= adda_reg_read,
-	.reg_write	= adda_reg_write,
-	.fast_io	= true,
-	.max_register	= 24,
-};
-
 /* mixer controls */
 static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = {
 	SOC_DAPM_DOUBLE_R("DAC Playback Switch",
@@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev)
 		return PTR_ERR(base);
 	}
 
-	regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg);
+	regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base);
 	if (IS_ERR(regmap)) {
 		dev_err(&pdev->dev, "Failed to create regmap\n");
 		return PTR_ERR(regmap);
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index fb37dd927e33..522a72fde78d 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -24,6 +24,7 @@
 #include <linux/io.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
+#include <linux/log2.h>
 
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -52,7 +53,6 @@
 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_INV		13
 #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV		9
 #define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV		6
-#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16		(1 << 6)
 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ		4
 #define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_16		(1 << 4)
 #define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT		2
@@ -300,12 +300,23 @@ static u8 sun8i_codec_get_bclk_div(struct sun8i_codec *scodec,
 	return best_val;
 }
 
+static int sun8i_codec_get_lrck_div(unsigned int channels,
+				    unsigned int word_size)
+{
+	unsigned int div = word_size * channels;
+
+	if (div < 16 || div > 256)
+		return -EINVAL;
+
+	return ilog2(div) - 4;
+}
+
 static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
 	struct sun8i_codec *scodec = snd_soc_component_get_drvdata(dai->component);
-	int sample_rate;
+	int sample_rate, lrck_div;
 	u8 bclk_div;
 
 	/*
@@ -321,9 +332,14 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream,
 			   SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK,
 			   bclk_div << SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV);
 
+	lrck_div = sun8i_codec_get_lrck_div(params_channels(params),
+					    params_physical_width(params));
+	if (lrck_div < 0)
+		return lrck_div;
+
 	regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL,
 			   SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK,
-			   SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_16);
+			   lrck_div << SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV);
 
 	sample_rate = sun8i_codec_get_hw_rate(params);
 	if (sample_rate < 0)
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index 45a4aa9d2a47..901457da25ec 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing/invalid\n");
 		ret = -EINVAL;
-		goto err;
+		goto err_put_codec_of_node;
 	}
 
 	tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
-		goto err;
+		goto err_put_cpu_of_node;
 
 	ret = snd_soc_register_card(card);
 	if (ret) {
@@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
 
 err_fini_utils:
 	tegra_asoc_utils_fini(&machine->util_data);
+err_put_cpu_of_node:
+	of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+	tegra_sgtl5000_dai.cpu_of_node = NULL;
+	tegra_sgtl5000_dai.platform_of_node = NULL;
+err_put_codec_of_node:
+	of_node_put(tegra_sgtl5000_dai.codec_of_node);
+	tegra_sgtl5000_dai.codec_of_node = NULL;
 err:
 	return ret;
 }
@@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
 
 	tegra_asoc_utils_fini(&machine->util_data);
 
+	of_node_put(tegra_sgtl5000_dai.cpu_of_node);
+	tegra_sgtl5000_dai.cpu_of_node = NULL;
+	tegra_sgtl5000_dai.platform_of_node = NULL;
+	of_node_put(tegra_sgtl5000_dai.codec_of_node);
+	tegra_sgtl5000_dai.codec_of_node = NULL;
+
 	return ret;
 }
 
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e2ad00e3cae1..1cfca698ae4b 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
 	if (err < 0)
 		return err;
 
-	return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
+	return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component,
 					  &txx9aclc_ac97_dai, 1);
 }
 
 static int txx9aclc_ac97_dev_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_set_ac97_ops(NULL);
 	return 0;
 }